Adding upstream version 115.7.0esr.upstream/115.7.0esr upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
author: Daniel Baumann <daniel.baumann@progress-linux.org> 2024-04-07 19:33:14 +0000
committer: Daniel Baumann <daniel.baumann@progress-linux.org> 2024-04-07 19:33:14 +0000
commit: 36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree: 105e8c98ddea1c1e4784a60a5a6410fa416be2de /dom/media/webrtc/tests
parent: Initial commit. (diff)
download: firefox-esr-upstream.tar.xz
firefox-esr-upstream.zip
297 files changed, 29234 insertions, 0 deletions
diff --git a/dom/media/webrtc/tests/crashtests/1770075.html b/dom/media/webrtc/tests/crashtests/1770075.html
new file mode 100644
index 0000000000..4d451216bd
--- /dev/null
+++ b/dom/media/webrtc/tests/crashtests/1770075.html
@@ -0,0 +1,8 @@
+<script>
+window.addEventListener('load', () => {
+  let a = new RTCPeerConnection({}, {})
+  a.createOffer({'offerToReceiveVideo': true})
+  let b = new WeakRef(a.getTransceivers()[0])
+  setTimeout("self.close()", 200)
+})
+</script>
diff --git a/dom/media/webrtc/tests/crashtests/1789908.html b/dom/media/webrtc/tests/crashtests/1789908.html
new file mode 100644
index 0000000000..3d58d3dc6b
--- /dev/null
+++ b/dom/media/webrtc/tests/crashtests/1789908.html
@@ -0,0 +1,25 @@
+<script>
+window.addEventListener('load', () => {
+  const sdp = `v=0
+o=mozilla...THIS_IS_SDPARTA-99.0 4978061689314146455 0 IN IP4 0.0.0.0
+s=-
+t=0 0
+a=fingerprint:sha-256 1D:E5:0C:97:18:43:38:3D:FF:7D:6A:BF:E3:AC:CA:70:AB:53:5A:35:95:92:4F:98:86:61:CA:5D:D5:9D:5E:41
+a=group:BUNDLE 0
+a=ice-options:trickle
+a=msid-semantic:WMS *
+m=video 9 UDP/TLS/RTP/SAVPF 120
+c=IN IP4 0.0.0.0
+a=fmtp:120 max-fs=12288;max-fr=60
+a=ice-pwd:c3a5e05023a8c38f671aef91ed1802d6
+a=ice-ufrag:91e4526d
+a=setup:actpass
+
+a=rtpmap:120 VP8/90000
+`;
+
+  let a = new RTCPeerConnection()
+  a.setRemoteDescription({sdp, type: "offer"});
+  setTimeout("self.close()", 200)
+})
+</script>
diff --git a/dom/media/webrtc/tests/crashtests/1799168.html b/dom/media/webrtc/tests/crashtests/1799168.html
new file mode 100644
index 0000000000..6c5c9db237
--- /dev/null
+++ b/dom/media/webrtc/tests/crashtests/1799168.html
@@ -0,0 +1,16 @@
+<script>
+window.addEventListener('load', async () => {
+  const offerer = new RTCPeerConnection();
+  const answerer = new RTCPeerConnection();
+  offerer.addTransceiver('audio');
+  await offerer.setLocalDescription();
+  await answerer.setRemoteDescription(offerer.localDescription);
+  const answer = await answerer.createAnswer();
+  await offerer.setRemoteDescription(answer);
+  // relay candidate with TCP!
+  const candidate = 'candidate:3 1 tcp 18087935 20.253.151.225 3478 typ relay raddr 10.0.48.153 rport 3478 tcptype passive';
+  await offerer.addIceCandidate({candidate, sdpMLineIndex: 0});
+  await new Promise(r => setTimeout(r, 2000));
+  self.close();
+})
+</script>
diff --git a/dom/media/webrtc/tests/crashtests/1816708.html b/dom/media/webrtc/tests/crashtests/1816708.html
new file mode 100644
index 0000000000..c7ba824041
--- /dev/null
+++ b/dom/media/webrtc/tests/crashtests/1816708.html
@@ -0,0 +1,21 @@
+<!DOCTYPE html>
+<html class="reftest-wait">
+<head>
+<script>
+document.addEventListener('DOMContentLoaded', async () => {
+  const peer = new RTCPeerConnection()
+  const stream = await navigator.mediaDevices.getUserMedia({
+    video: true,
+    audio: true,
+    fake: true,
+    peerIdentity: 'name',
+  })
+  stream.getTracks().forEach((track) => peer.addTrack(track, stream))
+  const offer = await peer.createOffer({})
+  await peer.setLocalDescription(offer)
+  await peer.setRemoteDescription(offer)
+  document.documentElement.removeAttribute("class");
+})
+</script>
+</head>
+</html>
diff --git a/dom/media/webrtc/tests/crashtests/1821477.html b/dom/media/webrtc/tests/crashtests/1821477.html
new file mode 100644
index 0000000000..c37bd6bd02
--- /dev/null
+++ b/dom/media/webrtc/tests/crashtests/1821477.html
@@ -0,0 +1,16 @@
+<html class="reftest-wait">
+<script>
+document.addEventListener("DOMContentLoaded", async () => {
+  SpecialPowers.wrap(document).notifyUserGestureActivation();
+  try {
+    (await navigator.mediaDevices.getDisplayMedia({
+      "video": {
+        "frameRate": 2147483647,
+      },
+    })).stop();
+  } finally {
+    document.documentElement.removeAttribute("class");
+  }
+});
+</script>
+</html>
diff --git a/dom/media/webrtc/tests/crashtests/crashtests.list b/dom/media/webrtc/tests/crashtests/crashtests.list
new file mode 100644
index 0000000000..64434b28e2
--- /dev/null
+++ b/dom/media/webrtc/tests/crashtests/crashtests.list
@@ -0,0 +1,7 @@
+defaults pref(media.navigator.permission.disabled,true) pref(media.devices.insecure.enabled,true) pref(media.getusermedia.insecure.enabled,true)
+
+load 1770075.html
+load 1789908.html
+load 1799168.html
+load 1816708.html
+skip-if(/^Windows\x20NT\x206\.1/.test(http.oscpu)) load 1821477.html
diff --git a/dom/media/webrtc/tests/fuzztests/moz.build b/dom/media/webrtc/tests/fuzztests/moz.build
new file mode 100644
index 0000000000..fef388e6c9
--- /dev/null
+++ b/dom/media/webrtc/tests/fuzztests/moz.build
@@ -0,0 +1,22 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+Library("FuzzingSdp")
+
+LOCAL_INCLUDES += [
+    "/dom/media/webrtc",
+    "/ipc/chromium/src",
+    "/media/webrtc",
+]
+
+# Add libFuzzer configuration directives
+include("/tools/fuzzing/libfuzzer-config.mozbuild")
+
+SOURCES += [
+    "sdp_parser_libfuzz.cpp",
+]
+
+FINAL_LIBRARY = "xul-gtest"
diff --git a/dom/media/webrtc/tests/fuzztests/sdp_parser_libfuzz.cpp b/dom/media/webrtc/tests/fuzztests/sdp_parser_libfuzz.cpp
new file mode 100644
index 0000000000..3451d6fd21
--- /dev/null
+++ b/dom/media/webrtc/tests/fuzztests/sdp_parser_libfuzz.cpp
@@ -0,0 +1,30 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include <string>
+
+#include "gtest/gtest.h"
+
+#include "FuzzingInterface.h"
+
+#include "sdp/SipccSdpParser.h"
+
+using namespace mozilla;
+
+static mozilla::UniquePtr<SdpParser::Results> sdpPtr;
+static SipccSdpParser mParser;
+
+int FuzzingInitSdpParser(int* argc, char*** argv) { return 0; }
+
+static int RunSdpParserFuzzing(const uint8_t* data, size_t size) {
+  std::string message(reinterpret_cast<const char*>(data), size);
+
+  sdpPtr = mParser.Parse(message);
+
+  return 0;
+}
+
+MOZ_FUZZING_INTERFACE_RAW(FuzzingInitSdpParser, RunSdpParserFuzzing, SdpParser);
diff --git a/dom/media/webrtc/tests/mochitests/NetworkPreparationChromeScript.js b/dom/media/webrtc/tests/mochitests/NetworkPreparationChromeScript.js
new file mode 100644
index 0000000000..d3872f1519
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/NetworkPreparationChromeScript.js
@@ -0,0 +1,43 @@
+/* Any copyright is dedicated to the Public Domain.
+   http://creativecommons.org/publicdomain/zero/1.0/ */
+
+var browser = Services.wm.getMostRecentWindow("navigator:browser");
+var connection = browser.navigator.mozMobileConnections[0];
+
+// provide a fake APN and enable data connection.
+// enable 3G radio
+function enableRadio() {
+  if (connection.radioState !== "enabled") {
+    connection.setRadioEnabled(true);
+  }
+}
+
+// disable 3G radio
+function disableRadio() {
+  if (connection.radioState === "enabled") {
+    connection.setRadioEnabled(false);
+  }
+}
+
+addMessageListener("prepare-network", function (message) {
+  connection.addEventListener("datachange", function onDataChange() {
+    if (connection.data.connected) {
+      connection.removeEventListener("datachange", onDataChange);
+      Services.prefs.setIntPref("network.proxy.type", 2);
+      sendAsyncMessage("network-ready", true);
+    }
+  });
+
+  enableRadio();
+});
+
+addMessageListener("network-cleanup", function (message) {
+  connection.addEventListener("datachange", function onDataChange() {
+    if (!connection.data.connected) {
+      connection.removeEventListener("datachange", onDataChange);
+      Services.prefs.setIntPref("network.proxy.type", 2);
+      sendAsyncMessage("network-disabled", true);
+    }
+  });
+  disableRadio();
+});
diff --git a/dom/media/webrtc/tests/mochitests/addTurnsSelfsignedCert.js b/dom/media/webrtc/tests/mochitests/addTurnsSelfsignedCert.js
new file mode 100644
index 0000000000..1e8be3a397
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/addTurnsSelfsignedCert.js
@@ -0,0 +1,32 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/* eslint-env mozilla/chrome-script */
+
+"use strict";
+
+// This is only usable from the parent process, even for doing simple stuff like
+// serializing a cert.
+var gCertMaker = Cc["@mozilla.org/security/x509certdb;1"].getService(
+  Ci.nsIX509CertDB
+);
+
+var gCertOverrides = Cc["@mozilla.org/security/certoverride;1"].getService(
+  Ci.nsICertOverrideService
+);
+
+addMessageListener("add-turns-certs", certs => {
+  var port = 5349;
+  certs.forEach(certDescription => {
+    var cert = gCertMaker.constructX509FromBase64(certDescription.cert);
+    gCertOverrides.rememberValidityOverride(
+      certDescription.hostname,
+      port,
+      {},
+      cert,
+      false
+    );
+  });
+  sendAsyncMessage("certs-added");
+});
diff --git a/dom/media/webrtc/tests/mochitests/blacksilence.js b/dom/media/webrtc/tests/mochitests/blacksilence.js
new file mode 100644
index 0000000000..5ea35f8a7f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/blacksilence.js
@@ -0,0 +1,134 @@
+(function (global) {
+  "use strict";
+
+  // an invertible check on the condition.
+  // if the constraint is applied, then the check is direct
+  // if not applied, then the result should be reversed
+  function check(constraintApplied, condition, message) {
+    var good = constraintApplied ? condition : !condition;
+    message =
+      (constraintApplied ? "with" : "without") +
+      " constraint: should " +
+      (constraintApplied ? "" : "not ") +
+      message +
+      " = " +
+      (good ? "OK" : "waiting...");
+    info(message);
+    return good;
+  }
+
+  function mkElement(type) {
+    // This makes an unattached element.
+    // It's not rendered to save the cycles that costs on b2g emulator
+    // and it gets dropped (and GC'd) when the test is done.
+    var e = document.createElement(type);
+    e.width = 32;
+    e.height = 24;
+    document.getElementById("display").appendChild(e);
+    return e;
+  }
+
+  // Runs checkFunc until it reports success.
+  // This is kludgy, but you have to wait for media to start flowing, and it
+  // can't be any old media, it has to include real data, for which we have no
+  // reliable signals to use as a trigger.
+  function periodicCheck(checkFunc) {
+    var resolve;
+    var done = false;
+    // This returns a function so that we create 10 closures in the loop, not
+    // one; and so that the timers don't all start straight away
+    var waitAndCheck = counter => () => {
+      if (done) {
+        return Promise.resolve();
+      }
+      return new Promise(r => setTimeout(r, 200 << counter)).then(() => {
+        if (checkFunc()) {
+          done = true;
+          resolve();
+        }
+      });
+    };
+
+    var chain = Promise.resolve();
+    for (var i = 0; i < 10; ++i) {
+      chain = chain.then(waitAndCheck(i));
+    }
+    return new Promise(r => (resolve = r));
+  }
+
+  function isSilence(audioData) {
+    var silence = true;
+    for (var i = 0; i < audioData.length; ++i) {
+      if (audioData[i] !== 128) {
+        silence = false;
+      }
+    }
+    return silence;
+  }
+
+  function checkAudio(constraintApplied, stream) {
+    var audio = mkElement("audio");
+    audio.srcObject = stream;
+    audio.play();
+
+    var context = new AudioContext();
+    var source = context.createMediaStreamSource(stream);
+    var analyser = context.createAnalyser();
+    source.connect(analyser);
+    analyser.connect(context.destination);
+
+    return periodicCheck(() => {
+      var sampleCount = analyser.frequencyBinCount;
+      info("got some audio samples: " + sampleCount);
+      var buffer = new Uint8Array(sampleCount);
+      analyser.getByteTimeDomainData(buffer);
+
+      var silent = check(
+        constraintApplied,
+        isSilence(buffer),
+        "be silence for audio"
+      );
+      return sampleCount > 0 && silent;
+    }).then(() => {
+      source.disconnect();
+      analyser.disconnect();
+      audio.pause();
+      ok(true, "audio is " + (constraintApplied ? "" : "not ") + "silent");
+    });
+  }
+
+  function checkVideo(constraintApplied, stream) {
+    var video = mkElement("video");
+    video.srcObject = stream;
+    video.play();
+
+    return periodicCheck(() => {
+      try {
+        var canvas = mkElement("canvas");
+        var ctx = canvas.getContext("2d");
+        // Have to guard drawImage with the try as well, due to bug 879717. If
+        // we get an error, this round fails, but that failure is usually just
+        // transitory.
+        ctx.drawImage(video, 0, 0);
+        ctx.getImageData(0, 0, 1, 1);
+        return check(
+          constraintApplied,
+          false,
+          "throw on getImageData for video"
+        );
+      } catch (e) {
+        return check(
+          constraintApplied,
+          e.name === "SecurityError",
+          "get a security error: " + e.name
+        );
+      }
+    }).then(() => {
+      video.pause();
+      ok(true, "video is " + (constraintApplied ? "" : "not ") + "protected");
+    });
+  }
+
+  global.audioIsSilence = checkAudio;
+  global.videoIsBlack = checkVideo;
+})(this);
diff --git a/dom/media/webrtc/tests/mochitests/dataChannel.js b/dom/media/webrtc/tests/mochitests/dataChannel.js
new file mode 100644
index 0000000000..eac52f96ab
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/dataChannel.js
@@ -0,0 +1,352 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/**
+ * Returns the contents of a blob as text
+ *
+ * @param {Blob} blob
+          The blob to retrieve the contents from
+ */
+function getBlobContent(blob) {
+  return new Promise(resolve => {
+    var reader = new FileReader();
+    // Listen for 'onloadend' which will always be called after a success or failure
+    reader.onloadend = event => resolve(event.target.result);
+    reader.readAsText(blob);
+  });
+}
+
+var commandsCreateDataChannel = [
+  function PC_REMOTE_EXPECT_DATA_CHANNEL(test) {
+    test.pcRemote.expectDataChannel();
+  },
+
+  function PC_LOCAL_CREATE_DATA_CHANNEL(test) {
+    var channel = test.pcLocal.createDataChannel({});
+    is(channel.binaryType, "blob", channel + " is of binary type 'blob'");
+
+    is(
+      test.pcLocal.signalingState,
+      STABLE,
+      "Create datachannel does not change signaling state"
+    );
+    return test.pcLocal.observedNegotiationNeeded;
+  },
+];
+
+var commandsWaitForDataChannel = [
+  function PC_LOCAL_VERIFY_DATA_CHANNEL_STATE(test) {
+    return test.pcLocal.dataChannels[0].opened;
+  },
+
+  function PC_REMOTE_VERIFY_DATA_CHANNEL_STATE(test) {
+    return test.pcRemote.nextDataChannel.then(channel => channel.opened);
+  },
+];
+
+var commandsCheckDataChannel = [
+  function SEND_MESSAGE(test) {
+    var message = "Lorem ipsum dolor sit amet";
+
+    info("Sending message:" + message);
+    return test.send(message).then(result => {
+      is(
+        result.data,
+        message,
+        "Message correctly transmitted from pcLocal to pcRemote."
+      );
+    });
+  },
+
+  function SEND_BLOB(test) {
+    var contents = "At vero eos et accusam et justo duo dolores et ea rebum.";
+    var blob = new Blob([contents], { type: "text/plain" });
+
+    info("Sending blob");
+    return test
+      .send(blob)
+      .then(result => {
+        ok(result.data instanceof Blob, "Received data is of instance Blob");
+        is(result.data.size, blob.size, "Received data has the correct size.");
+
+        return getBlobContent(result.data);
+      })
+      .then(recv_contents =>
+        is(recv_contents, contents, "Received data has the correct content.")
+      );
+  },
+
+  function CREATE_SECOND_DATA_CHANNEL(test) {
+    return test.createDataChannel({}).then(result => {
+      is(
+        result.remote.binaryType,
+        "blob",
+        "remote data channel is of binary type 'blob'"
+      );
+    });
+  },
+
+  function SEND_MESSAGE_THROUGH_LAST_OPENED_CHANNEL(test) {
+    var channels = test.pcRemote.dataChannels;
+    var message = "I am the Omega";
+
+    info("Sending message:" + message);
+    return test.send(message).then(result => {
+      is(
+        channels.indexOf(result.channel),
+        channels.length - 1,
+        "Last channel used"
+      );
+      is(result.data, message, "Received message has the correct content.");
+    });
+  },
+
+  function SEND_MESSAGE_THROUGH_FIRST_CHANNEL(test) {
+    var message = "Message through 1st channel";
+    var options = {
+      sourceChannel: test.pcLocal.dataChannels[0],
+      targetChannel: test.pcRemote.dataChannels[0],
+    };
+
+    info("Sending message:" + message);
+    return test.send(message, options).then(result => {
+      is(
+        test.pcRemote.dataChannels.indexOf(result.channel),
+        0,
+        "1st channel used"
+      );
+      is(result.data, message, "Received message has the correct content.");
+    });
+  },
+
+  function SEND_MESSAGE_BACK_THROUGH_FIRST_CHANNEL(test) {
+    var message = "Return a message also through 1st channel";
+    var options = {
+      sourceChannel: test.pcRemote.dataChannels[0],
+      targetChannel: test.pcLocal.dataChannels[0],
+    };
+
+    info("Sending message:" + message);
+    return test.send(message, options).then(result => {
+      is(
+        test.pcLocal.dataChannels.indexOf(result.channel),
+        0,
+        "1st channel used"
+      );
+      is(result.data, message, "Return message has the correct content.");
+    });
+  },
+
+  function CREATE_NEGOTIATED_DATA_CHANNEL_MAX_RETRANSMITS(test) {
+    var options = {
+      negotiated: true,
+      id: 5,
+      protocol: "foo/bar",
+      ordered: false,
+      maxRetransmits: 500,
+    };
+    return test.createDataChannel(options).then(result => {
+      is(
+        result.local.binaryType,
+        "blob",
+        result.remote + " is of binary type 'blob'"
+      );
+      is(
+        result.local.id,
+        options.id,
+        result.local + " id is:" + result.local.id
+      );
+      is(
+        result.local.protocol,
+        options.protocol,
+        result.local + " protocol is:" + result.local.protocol
+      );
+      is(
+        result.local.reliable,
+        false,
+        result.local + " reliable is:" + result.local.reliable
+      );
+      is(
+        result.local.ordered,
+        options.ordered,
+        result.local + " ordered is:" + result.local.ordered
+      );
+      is(
+        result.local.maxRetransmits,
+        options.maxRetransmits,
+        result.local + " maxRetransmits is:" + result.local.maxRetransmits
+      );
+      is(
+        result.local.maxPacketLifeTime,
+        null,
+        result.local + " maxPacketLifeTime is:" + result.local.maxPacketLifeTime
+      );
+
+      is(
+        result.remote.binaryType,
+        "blob",
+        result.remote + " is of binary type 'blob'"
+      );
+      is(
+        result.remote.id,
+        options.id,
+        result.remote + " id is:" + result.remote.id
+      );
+      is(
+        result.remote.protocol,
+        options.protocol,
+        result.remote + " protocol is:" + result.remote.protocol
+      );
+      is(
+        result.remote.reliable,
+        false,
+        result.remote + " reliable is:" + result.remote.reliable
+      );
+      is(
+        result.remote.ordered,
+        options.ordered,
+        result.remote + " ordered is:" + result.remote.ordered
+      );
+      is(
+        result.remote.maxRetransmits,
+        options.maxRetransmits,
+        result.remote + " maxRetransmits is:" + result.remote.maxRetransmits
+      );
+      is(
+        result.remote.maxPacketLifeTime,
+        null,
+        result.remote +
+          " maxPacketLifeTime is:" +
+          result.remote.maxPacketLifeTime
+      );
+    });
+  },
+
+  function SEND_MESSAGE_THROUGH_LAST_OPENED_CHANNEL2(test) {
+    var channels = test.pcRemote.dataChannels;
+    var message = "I am the walrus; Goo goo g'joob";
+
+    info("Sending message:" + message);
+    return test.send(message).then(result => {
+      is(
+        channels.indexOf(result.channel),
+        channels.length - 1,
+        "Last channel used"
+      );
+      is(result.data, message, "Received message has the correct content.");
+    });
+  },
+
+  function CREATE_NEGOTIATED_DATA_CHANNEL_MAX_PACKET_LIFE_TIME(test) {
+    var options = {
+      ordered: false,
+      maxPacketLifeTime: 10,
+    };
+    return test.createDataChannel(options).then(result => {
+      is(
+        result.local.binaryType,
+        "blob",
+        result.local + " is of binary type 'blob'"
+      );
+      is(
+        result.local.protocol,
+        "",
+        result.local + " protocol is:" + result.local.protocol
+      );
+      is(
+        result.local.reliable,
+        false,
+        result.local + " reliable is:" + result.local.reliable
+      );
+      is(
+        result.local.ordered,
+        options.ordered,
+        result.local + " ordered is:" + result.local.ordered
+      );
+      is(
+        result.local.maxRetransmits,
+        null,
+        result.local + " maxRetransmits is:" + result.local.maxRetransmits
+      );
+      is(
+        result.local.maxPacketLifeTime,
+        options.maxPacketLifeTime,
+        result.local + " maxPacketLifeTime is:" + result.local.maxPacketLifeTime
+      );
+
+      is(
+        result.remote.binaryType,
+        "blob",
+        result.remote + " is of binary type 'blob'"
+      );
+      is(
+        result.remote.protocol,
+        "",
+        result.remote + " protocol is:" + result.remote.protocol
+      );
+      is(
+        result.remote.reliable,
+        false,
+        result.remote + " reliable is:" + result.remote.reliable
+      );
+      is(
+        result.remote.ordered,
+        options.ordered,
+        result.remote + " ordered is:" + result.remote.ordered
+      );
+      is(
+        result.remote.maxRetransmits,
+        null,
+        result.remote + " maxRetransmits is:" + result.remote.maxRetransmits
+      );
+      is(
+        result.remote.maxPacketLifeTime,
+        options.maxPacketLifeTime,
+        result.remote +
+          " maxPacketLifeTime is:" +
+          result.remote.maxPacketLifeTime
+      );
+    });
+  },
+
+  function SEND_MESSAGE_THROUGH_LAST_OPENED_CHANNEL3(test) {
+    var channels = test.pcRemote.dataChannels;
+    var message = "Nice to see you working maxPacketLifeTime";
+
+    info("Sending message:" + message);
+    return test.send(message).then(result => {
+      is(
+        channels.indexOf(result.channel),
+        channels.length - 1,
+        "Last channel used"
+      );
+      is(result.data, message, "Received message has the correct content.");
+    });
+  },
+];
+
+var commandsCheckLargeXfer = [
+  function SEND_BIG_BUFFER(test) {
+    var size = 2 * 1024 * 1024; // SCTP internal buffer is now 1MB, so use 2MB to ensure the buffer gets full
+    var buffer = new ArrayBuffer(size);
+    // note: type received is always blob for binary data
+    var options = {};
+    options.bufferedAmountLowThreshold = 64 * 1024;
+    info("Sending arraybuffer");
+    return test.send(buffer, options).then(result => {
+      ok(result.data instanceof Blob, "Received data is of instance Blob");
+      is(result.data.size, size, "Received data has the correct size.");
+    });
+  },
+];
+
+function addInitialDataChannel(chain) {
+  chain.insertBefore("PC_LOCAL_CREATE_OFFER", commandsCreateDataChannel);
+  chain.insertBefore(
+    "PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+    commandsWaitForDataChannel
+  );
+  chain.removeAfter("PC_REMOTE_CHECK_ICE_CONNECTIONS");
+  chain.append(commandsCheckDataChannel);
+}
diff --git a/dom/media/webrtc/tests/mochitests/head.js b/dom/media/webrtc/tests/mochitests/head.js
new file mode 100644
index 0000000000..7c9f6d52de
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/head.js
@@ -0,0 +1,1445 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+"use strict";
+
+var Cc = SpecialPowers.Cc;
+var Ci = SpecialPowers.Ci;
+
+// Specifies if we want fake audio streams for this run
+let WANT_FAKE_AUDIO = true;
+// Specifies if we want fake video streams for this run
+let WANT_FAKE_VIDEO = true;
+let TEST_AUDIO_FREQ = 1000;
+
+/**
+ * Reads the current values of preferences affecting fake and loopback devices
+ * and sets the WANT_FAKE_AUDIO and WANT_FAKE_VIDEO gloabals appropriately.
+ */
+function updateConfigFromFakeAndLoopbackPrefs() {
+  let audioDevice = SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (audioDevice) {
+    WANT_FAKE_AUDIO = false;
+    dump("TEST DEVICES: Got loopback audio: " + audioDevice + "\n");
+  } else {
+    WANT_FAKE_AUDIO = true;
+    dump(
+      "TEST DEVICES: No test device found in media.audio_loopback_dev, using fake audio streams.\n"
+    );
+  }
+  let videoDevice = SpecialPowers.getCharPref("media.video_loopback_dev", "");
+  if (videoDevice) {
+    WANT_FAKE_VIDEO = false;
+    dump("TEST DEVICES: Got loopback video: " + videoDevice + "\n");
+  } else {
+    WANT_FAKE_VIDEO = true;
+    dump(
+      "TEST DEVICES: No test device found in media.video_loopback_dev, using fake video streams.\n"
+    );
+  }
+}
+
+updateConfigFromFakeAndLoopbackPrefs();
+
+/**
+ *  Global flag to skip LoopbackTone
+ */
+let DISABLE_LOOPBACK_TONE = false;
+/**
+ * Helper class to setup a sine tone of a given frequency.
+ */
+class LoopbackTone {
+  constructor(audioContext, frequency) {
+    if (!audioContext) {
+      throw new Error("You must provide a valid AudioContext");
+    }
+    this.oscNode = audioContext.createOscillator();
+    var gainNode = audioContext.createGain();
+    gainNode.gain.value = 0.5;
+    this.oscNode.connect(gainNode);
+    gainNode.connect(audioContext.destination);
+    this.changeFrequency(frequency);
+  }
+
+  // Method should be used when WANT_FAKE_AUDIO is false.
+  start() {
+    if (!this.oscNode) {
+      throw new Error("Attempt to start a stopped LoopbackTone");
+    }
+    info(`Start loopback tone at ${this.oscNode.frequency.value}`);
+    this.oscNode.start();
+  }
+
+  // Change the frequency of the tone. It can be used after start.
+  // Frequency will change on the fly. No need to stop and create a new instance.
+  changeFrequency(frequency) {
+    if (!this.oscNode) {
+      throw new Error("Attempt to change frequency on a stopped LoopbackTone");
+    }
+    this.oscNode.frequency.value = frequency;
+  }
+
+  stop() {
+    if (!this.oscNode) {
+      throw new Error("Attempt to stop a stopped LoopbackTone");
+    }
+    this.oscNode.stop();
+    this.oscNode = null;
+  }
+}
+// Object that holds the default loopback tone.
+var DefaultLoopbackTone = null;
+
+/**
+ * This class provides helpers around analysing the audio content in a stream
+ * using WebAudio AnalyserNodes.
+ *
+ * @constructor
+ * @param {object} stream
+ *                 A MediaStream object whose audio track we shall analyse.
+ */
+function AudioStreamAnalyser(ac, stream) {
+  this.audioContext = ac;
+  this.stream = stream;
+  this.sourceNodes = [];
+  this.analyser = this.audioContext.createAnalyser();
+  // Setting values lower than default for speedier testing on emulators
+  this.analyser.smoothingTimeConstant = 0.2;
+  this.analyser.fftSize = 1024;
+  this.connectTrack = t => {
+    let source = this.audioContext.createMediaStreamSource(
+      new MediaStream([t])
+    );
+    this.sourceNodes.push(source);
+    source.connect(this.analyser);
+  };
+  this.stream.getAudioTracks().forEach(t => this.connectTrack(t));
+  this.onaddtrack = ev => this.connectTrack(ev.track);
+  this.stream.addEventListener("addtrack", this.onaddtrack);
+  this.data = new Uint8Array(this.analyser.frequencyBinCount);
+}
+
+AudioStreamAnalyser.prototype = {
+  /**
+   * Get an array of frequency domain data for our stream's audio track.
+   *
+   * @returns {array} A Uint8Array containing the frequency domain data.
+   */
+  getByteFrequencyData() {
+    this.analyser.getByteFrequencyData(this.data);
+    return this.data;
+  },
+
+  /**
+   * Append a canvas to the DOM where the frequency data are drawn.
+   * Useful to debug tests.
+   */
+  enableDebugCanvas() {
+    var cvs = (this.debugCanvas = document.createElement("canvas"));
+    const content = document.getElementById("content");
+    content.insertBefore(cvs, content.children[0]);
+
+    // Easy: 1px per bin
+    cvs.width = this.analyser.frequencyBinCount;
+    cvs.height = 128;
+    cvs.style.border = "1px solid red";
+
+    var c = cvs.getContext("2d");
+    c.fillStyle = "black";
+
+    var self = this;
+    function render() {
+      c.clearRect(0, 0, cvs.width, cvs.height);
+      var array = self.getByteFrequencyData();
+      for (var i = 0; i < array.length; i++) {
+        c.fillRect(i, cvs.height - array[i] / 2, 1, cvs.height);
+      }
+      if (!cvs.stopDrawing) {
+        requestAnimationFrame(render);
+      }
+    }
+    requestAnimationFrame(render);
+  },
+
+  /**
+   * Stop drawing of and remove the debug canvas from the DOM if it was
+   * previously added.
+   */
+  disableDebugCanvas() {
+    if (!this.debugCanvas || !this.debugCanvas.parentElement) {
+      return;
+    }
+
+    this.debugCanvas.stopDrawing = true;
+    this.debugCanvas.parentElement.removeChild(this.debugCanvas);
+  },
+
+  /**
+   * Disconnects the input stream from our internal analyser node.
+   * Call this to reduce main thread processing, mostly necessary on slow
+   * devices.
+   */
+  disconnect() {
+    this.disableDebugCanvas();
+    this.sourceNodes.forEach(n => n.disconnect());
+    this.sourceNodes = [];
+    this.stream.removeEventListener("addtrack", this.onaddtrack);
+  },
+
+  /**
+   * Return a Promise, that will be resolved when the function passed as
+   * argument, when called, returns true (meaning the analysis was a
+   * success). The promise is rejected if the cancel promise resolves first.
+   *
+   * @param {function} analysisFunction
+   *        A function that performs an analysis, and resolves with true if the
+   *        analysis was a success (i.e. it found what it was looking for)
+   * @param {promise} cancel
+   *        A promise that on resolving will reject the promise we returned.
+   */
+  async waitForAnalysisSuccess(
+    analysisFunction,
+    cancel = wait(60000, new Error("Audio analysis timed out"))
+  ) {
+    let aborted = false;
+    cancel.then(() => (aborted = true));
+
+    // We need to give the Analyser some time to start gathering data.
+    await wait(200);
+
+    do {
+      await new Promise(resolve => requestAnimationFrame(resolve));
+      if (aborted) {
+        throw await cancel;
+      }
+    } while (!analysisFunction(this.getByteFrequencyData()));
+  },
+
+  /**
+   * Return the FFT bin index for a given frequency.
+   *
+   * @param {double} frequency
+   *        The frequency for whicht to return the bin number.
+   * @returns {integer} the index of the bin in the FFT array.
+   */
+  binIndexForFrequency(frequency) {
+    return (
+      1 +
+      Math.round(
+        (frequency * this.analyser.fftSize) / this.audioContext.sampleRate
+      )
+    );
+  },
+
+  /**
+   * Reverse operation, get the frequency for a bin index.
+   *
+   * @param {integer} index an index in an FFT array
+   * @returns {double} the frequency for this bin
+   */
+  frequencyForBinIndex(index) {
+    return ((index - 1) * this.audioContext.sampleRate) / this.analyser.fftSize;
+  },
+};
+
+/**
+ * Creates a MediaStream with an audio track containing a sine tone at the
+ * given frequency.
+ *
+ * @param {AudioContext} ac
+ *        AudioContext in which to create the OscillatorNode backing the stream
+ * @param {double} frequency
+ *        The frequency in Hz of the generated sine tone
+ * @returns {MediaStream} the MediaStream containing sine tone audio track
+ */
+function createOscillatorStream(ac, frequency) {
+  var osc = ac.createOscillator();
+  osc.frequency.value = frequency;
+
+  var oscDest = ac.createMediaStreamDestination();
+  osc.connect(oscDest);
+  osc.start();
+  return oscDest.stream;
+}
+
+/**
+ * Create the necessary HTML elements for head and body as used by Mochitests
+ *
+ * @param {object} meta
+ *        Meta information of the test
+ * @param {string} meta.title
+ *        Description of the test
+ * @param {string} [meta.bug]
+ *        Bug the test was created for
+ * @param {boolean} [meta.visible=false]
+ *        Visibility of the media elements
+ */
+function realCreateHTML(meta) {
+  var test = document.getElementById("test");
+
+  // Create the head content
+  var elem = document.createElement("meta");
+  elem.setAttribute("charset", "utf-8");
+  document.head.appendChild(elem);
+
+  var title = document.createElement("title");
+  title.textContent = meta.title;
+  document.head.appendChild(title);
+
+  // Create the body content
+  var anchor = document.createElement("a");
+  anchor.textContent = meta.title;
+  if (meta.bug) {
+    anchor.setAttribute(
+      "href",
+      "https://bugzilla.mozilla.org/show_bug.cgi?id=" + meta.bug
+    );
+  } else {
+    anchor.setAttribute("target", "_blank");
+  }
+
+  document.body.insertBefore(anchor, test);
+
+  var display = document.createElement("p");
+  display.setAttribute("id", "display");
+  document.body.insertBefore(display, test);
+
+  var content = document.createElement("div");
+  content.setAttribute("id", "content");
+  content.style.display = meta.visible ? "block" : "none";
+  document.body.appendChild(content);
+}
+
+/**
+ * Creates an element of the given type, assigns the given id, sets the controls
+ * and autoplay attributes and adds it to the content node.
+ *
+ * @param {string} type
+ *        Defining if we should create an "audio" or "video" element
+ * @param {string} id
+ *        A string to use as the element id.
+ */
+function createMediaElement(type, id) {
+  const element = document.createElement(type);
+  element.setAttribute("id", id);
+  element.setAttribute("height", 100);
+  element.setAttribute("width", 150);
+  element.setAttribute("controls", "controls");
+  element.setAttribute("autoplay", "autoplay");
+  element.setAttribute("muted", "muted");
+  element.muted = true;
+  document.getElementById("content").appendChild(element);
+
+  return element;
+}
+
+/**
+ * Returns an existing element for the given track with the given idPrefix,
+ * as it was added by createMediaElementForTrack().
+ *
+ * @param {MediaStreamTrack} track
+ *        Track used as the element's source.
+ * @param {string} idPrefix
+ *        A string to use as the element id. The track id will also be appended.
+ */
+function getMediaElementForTrack(track, idPrefix) {
+  return document.getElementById(idPrefix + "_" + track.id);
+}
+
+/**
+ * Create a media element with a track as source and attach it to the content
+ * node.
+ *
+ * @param {MediaStreamTrack} track
+ *        Track for use as source.
+ * @param {string} idPrefix
+ *        A string to use as the element id. The track id will also be appended.
+ * @return {HTMLMediaElement} The created HTML media element
+ */
+function createMediaElementForTrack(track, idPrefix) {
+  const id = idPrefix + "_" + track.id;
+  const element = createMediaElement(track.kind, id);
+  element.srcObject = new MediaStream([track]);
+
+  return element;
+}
+
+/**
+ * Wrapper function for mediaDevices.getUserMedia used by some tests. Whether
+ * to use fake devices or not is now determined in pref further below instead.
+ *
+ * @param {Dictionary} constraints
+ *        The constraints for this mozGetUserMedia callback
+ */
+function getUserMedia(constraints) {
+  // Tests may have changed the values of prefs, so recheck
+  updateConfigFromFakeAndLoopbackPrefs();
+  if (
+    !WANT_FAKE_AUDIO &&
+    !constraints.fake &&
+    constraints.audio &&
+    !DISABLE_LOOPBACK_TONE
+  ) {
+    // Loopback device is configured, start the default loopback tone
+    if (!DefaultLoopbackTone) {
+      TEST_AUDIO_FREQ = 440;
+      DefaultLoopbackTone = new LoopbackTone(
+        new AudioContext(),
+        TEST_AUDIO_FREQ
+      );
+      DefaultLoopbackTone.start();
+    }
+    // Disable input processing mode when it's not explicity enabled.
+    // This is to avoid distortion of the loopback tone
+    constraints.audio = Object.assign(
+      {},
+      { autoGainControl: false },
+      { echoCancellation: false },
+      { noiseSuppression: false },
+      constraints.audio
+    );
+  } else {
+    // Fake device configured, ensure our test freq is correct.
+    TEST_AUDIO_FREQ = 1000;
+  }
+  info("Call getUserMedia for " + JSON.stringify(constraints));
+  return navigator.mediaDevices
+    .getUserMedia(constraints)
+    .then(stream => (checkMediaStreamTracks(constraints, stream), stream));
+}
+
+// These are the promises we use to track that the prerequisites for the test
+// are in place before running it.
+var setTestOptions;
+var testConfigured = new Promise(r => (setTestOptions = r));
+
+function pushPrefs(...p) {
+  return SpecialPowers.pushPrefEnv({ set: p });
+}
+
+async function withPrefs(prefs, func) {
+  await SpecialPowers.pushPrefEnv({ set: prefs });
+  try {
+    return await func();
+  } finally {
+    await SpecialPowers.popPrefEnv();
+  }
+}
+
+function setupEnvironment() {
+  var defaultMochitestPrefs = {
+    set: [
+      ["media.peerconnection.enabled", true],
+      ["media.peerconnection.identity.enabled", true],
+      ["media.peerconnection.identity.timeout", 120000],
+      ["media.peerconnection.ice.stun_client_maximum_transmits", 14],
+      ["media.peerconnection.ice.trickle_grace_period", 30000],
+      ["media.navigator.permission.disabled", true],
+      // If either fake audio or video is desired we enable fake streams.
+      // If loopback devices are set they will be chosen instead of fakes in gecko.
+      ["media.navigator.streams.fake", WANT_FAKE_AUDIO || WANT_FAKE_VIDEO],
+      ["media.getusermedia.audiocapture.enabled", true],
+      ["media.getusermedia.screensharing.enabled", true],
+      ["media.getusermedia.window.focus_source.enabled", false],
+      ["media.recorder.audio_node.enabled", true],
+      ["media.peerconnection.ice.obfuscate_host_addresses", false],
+      ["media.peerconnection.nat_simulator.filtering_type", ""],
+      ["media.peerconnection.nat_simulator.mapping_type", ""],
+      ["media.peerconnection.nat_simulator.block_tcp", false],
+      ["media.peerconnection.nat_simulator.block_udp", false],
+      ["media.peerconnection.nat_simulator.redirect_address", ""],
+      ["media.peerconnection.nat_simulator.redirect_targets", ""],
+    ],
+  };
+
+  if (navigator.userAgent.includes("Android")) {
+    defaultMochitestPrefs.set.push(
+      ["media.navigator.video.default_width", 320],
+      ["media.navigator.video.default_height", 240],
+      ["media.navigator.video.max_fr", 10],
+      ["media.autoplay.default", Ci.nsIAutoplay.ALLOWED]
+    );
+  }
+
+  // Platform codec prefs should be matched because fake H.264 GMP codec doesn't
+  // produce/consume real bitstreams. [TODO] remove after bug 1509012 is fixed.
+  const platformEncoderEnabled = SpecialPowers.getBoolPref(
+    "media.webrtc.platformencoder"
+  );
+  defaultMochitestPrefs.set.push([
+    "media.navigator.mediadatadecoder_h264_enabled",
+    platformEncoderEnabled,
+  ]);
+
+  // Running as a Mochitest.
+  SimpleTest.requestFlakyTimeout("WebRTC inherently depends on timeouts");
+  window.finish = () => SimpleTest.finish();
+  SpecialPowers.pushPrefEnv(defaultMochitestPrefs, setTestOptions);
+
+  // We don't care about waiting for this to complete, we just want to ensure
+  // that we don't build up a huge backlog of GC work.
+  SpecialPowers.exactGC();
+}
+
+// [TODO] remove after bug 1509012 is fixed.
+async function matchPlatformH264CodecPrefs() {
+  const hasHW264 =
+    SpecialPowers.getBoolPref("media.webrtc.platformencoder") &&
+    !SpecialPowers.getBoolPref("media.webrtc.platformencoder.sw_only") &&
+    (navigator.userAgent.includes("Android") ||
+      navigator.userAgent.includes("Mac OS X"));
+
+  await pushPrefs(
+    ["media.webrtc.platformencoder", hasHW264],
+    ["media.navigator.mediadatadecoder_h264_enabled", hasHW264]
+  );
+}
+
+async function runTestWhenReady(testFunc) {
+  setupEnvironment();
+  const options = await testConfigured;
+  try {
+    await testFunc(options);
+  } catch (e) {
+    ok(
+      false,
+      `Error executing test: ${e}
+${e.stack ? e.stack : ""}`
+    );
+  } finally {
+    SimpleTest.finish();
+  }
+}
+
+/**
+ * Checks that the media stream tracks have the expected amount of tracks
+ * with the correct attributes based on the type and constraints given.
+ *
+ * @param {Object} constraints specifies whether the stream should have
+ *                             audio, video, or both
+ * @param {String} type the type of media stream tracks being checked
+ * @param {sequence<MediaStreamTrack>} mediaStreamTracks the media stream
+ *                                     tracks being checked
+ */
+function checkMediaStreamTracksByType(constraints, type, mediaStreamTracks) {
+  if (constraints[type]) {
+    is(mediaStreamTracks.length, 1, "One " + type + " track shall be present");
+
+    if (mediaStreamTracks.length) {
+      is(mediaStreamTracks[0].kind, type, "Track kind should be " + type);
+      ok(mediaStreamTracks[0].id, "Track id should be defined");
+      ok(!mediaStreamTracks[0].muted, "Track should not be muted");
+    }
+  } else {
+    is(mediaStreamTracks.length, 0, "No " + type + " tracks shall be present");
+  }
+}
+
+/**
+ * Check that the given media stream contains the expected media stream
+ * tracks given the associated audio & video constraints provided.
+ *
+ * @param {Object} constraints specifies whether the stream should have
+ *                             audio, video, or both
+ * @param {MediaStream} mediaStream the media stream being checked
+ */
+function checkMediaStreamTracks(constraints, mediaStream) {
+  checkMediaStreamTracksByType(
+    constraints,
+    "audio",
+    mediaStream.getAudioTracks()
+  );
+  checkMediaStreamTracksByType(
+    constraints,
+    "video",
+    mediaStream.getVideoTracks()
+  );
+}
+
+/**
+ * Check that a media stream contains exactly a set of media stream tracks.
+ *
+ * @param {MediaStream} mediaStream the media stream being checked
+ * @param {Array} tracks the tracks that should exist in mediaStream
+ * @param {String} [message] an optional message to pass to asserts
+ */
+function checkMediaStreamContains(mediaStream, tracks, message) {
+  message = message ? message + ": " : "";
+  tracks.forEach(t =>
+    ok(
+      mediaStream.getTrackById(t.id),
+      message + "MediaStream " + mediaStream.id + " contains track " + t.id
+    )
+  );
+  is(
+    mediaStream.getTracks().length,
+    tracks.length,
+    message + "MediaStream " + mediaStream.id + " contains no extra tracks"
+  );
+}
+
+function checkMediaStreamCloneAgainstOriginal(clone, original) {
+  isnot(clone.id.length, 0, "Stream clone should have an id string");
+  isnot(clone, original, "Stream clone should be different from the original");
+  isnot(
+    clone.id,
+    original.id,
+    "Stream clone's id should be different from the original's"
+  );
+  is(
+    clone.getAudioTracks().length,
+    original.getAudioTracks().length,
+    "All audio tracks should get cloned"
+  );
+  is(
+    clone.getVideoTracks().length,
+    original.getVideoTracks().length,
+    "All video tracks should get cloned"
+  );
+  is(clone.active, original.active, "Active state should be preserved");
+  original
+    .getTracks()
+    .forEach(t =>
+      ok(!clone.getTrackById(t.id), "The clone's tracks should be originals")
+    );
+}
+
+function checkMediaStreamTrackCloneAgainstOriginal(clone, original) {
+  isnot(clone.id.length, 0, "Track clone should have an id string");
+  isnot(clone, original, "Track clone should be different from the original");
+  isnot(
+    clone.id,
+    original.id,
+    "Track clone's id should be different from the original's"
+  );
+  is(
+    clone.kind,
+    original.kind,
+    "Track clone's kind should be same as the original's"
+  );
+  is(
+    clone.enabled,
+    original.enabled,
+    "Track clone's kind should be same as the original's"
+  );
+  is(
+    clone.readyState,
+    original.readyState,
+    "Track clone's readyState should be same as the original's"
+  );
+  is(
+    clone.muted,
+    original.muted,
+    "Track clone's muted state should be same as the original's"
+  );
+}
+
+/*** Utility methods */
+
+/** The dreadful setTimeout, use sparingly */
+function wait(time, message) {
+  return new Promise(r => setTimeout(() => r(message), time));
+}
+
+/** The even more dreadful setInterval, use even more sparingly */
+function waitUntil(func, time) {
+  return new Promise(resolve => {
+    var interval = setInterval(() => {
+      if (func()) {
+        clearInterval(interval);
+        resolve();
+      }
+    }, time || 200);
+  });
+}
+
+/** Time out while waiting for a promise to get resolved or rejected. */
+var timeout = (promise, time, msg) =>
+  Promise.race([
+    promise,
+    wait(time).then(() => Promise.reject(new Error(msg))),
+  ]);
+
+/** Adds a |finally| function to a promise whose argument is invoked whether the
+ * promise is resolved or rejected, and that does not interfere with chaining.*/
+var addFinallyToPromise = promise => {
+  promise.finally = func => {
+    return promise.then(
+      result => {
+        func();
+        return Promise.resolve(result);
+      },
+      error => {
+        func();
+        return Promise.reject(error);
+      }
+    );
+  };
+  return promise;
+};
+
+/** Use event listener to call passed-in function on fire until it returns true */
+var listenUntil = (target, eventName, onFire) => {
+  return new Promise(resolve =>
+    target.addEventListener(eventName, function callback(event) {
+      var result = onFire(event);
+      if (result) {
+        target.removeEventListener(eventName, callback);
+        resolve(result);
+      }
+    })
+  );
+};
+
+/* Test that a function throws the right error */
+function mustThrowWith(msg, reason, f) {
+  try {
+    f();
+    ok(false, msg + " must throw");
+  } catch (e) {
+    is(e.name, reason, msg + " must throw: " + e.message);
+  }
+}
+
+/* Get a dummy audio track */
+function getSilentTrack() {
+  let ctx = new AudioContext(),
+    oscillator = ctx.createOscillator();
+  let dst = oscillator.connect(ctx.createMediaStreamDestination());
+  oscillator.start();
+  return Object.assign(dst.stream.getAudioTracks()[0], { enabled: false });
+}
+
+function getBlackTrack({ width = 640, height = 480 } = {}) {
+  let canvas = Object.assign(document.createElement("canvas"), {
+    width,
+    height,
+  });
+  canvas.getContext("2d").fillRect(0, 0, width, height);
+  let stream = canvas.captureStream();
+  return Object.assign(stream.getVideoTracks()[0], { enabled: false });
+}
+
+/*** Test control flow methods */
+
+/**
+ * Generates a callback function fired only under unexpected circumstances
+ * while running the tests. The generated function kills off the test as well
+ * gracefully.
+ *
+ * @param {String} [message]
+ *        An optional message to show if no object gets passed into the
+ *        generated callback method.
+ */
+function generateErrorCallback(message) {
+  var stack = new Error().stack.split("\n");
+  stack.shift(); // Don't include this instantiation frame
+
+  /**
+   * @param {object} aObj
+   *        The object fired back from the callback
+   */
+  return aObj => {
+    if (aObj) {
+      if (aObj.name && aObj.message) {
+        ok(
+          false,
+          "Unexpected callback for '" +
+            aObj.name +
+            "' with message = '" +
+            aObj.message +
+            "' at " +
+            JSON.stringify(stack)
+        );
+      } else {
+        ok(
+          false,
+          "Unexpected callback with = '" +
+            aObj +
+            "' at: " +
+            JSON.stringify(stack)
+        );
+      }
+    } else {
+      ok(
+        false,
+        "Unexpected callback with message = '" +
+          message +
+          "' at: " +
+          JSON.stringify(stack)
+      );
+    }
+    throw new Error("Unexpected callback");
+  };
+}
+
+var unexpectedEventArrived;
+var rejectOnUnexpectedEvent = new Promise((x, reject) => {
+  unexpectedEventArrived = reject;
+});
+
+/**
+ * Generates a callback function fired only for unexpected events happening.
+ *
+ * @param {String} description
+          Description of the object for which the event has been fired
+ * @param {String} eventName
+          Name of the unexpected event
+ */
+function unexpectedEvent(message, eventName) {
+  var stack = new Error().stack.split("\n");
+  stack.shift(); // Don't include this instantiation frame
+
+  return e => {
+    var details =
+      "Unexpected event '" +
+      eventName +
+      "' fired with message = '" +
+      message +
+      "' at: " +
+      JSON.stringify(stack);
+    ok(false, details);
+    unexpectedEventArrived(new Error(details));
+  };
+}
+
+/**
+ * Implements the one-shot event pattern used throughout.  Each of the 'onxxx'
+ * attributes on the wrappers can be set with a custom handler.  Prior to the
+ * handler being set, if the event fires, it causes the test execution to halt.
+ * That handler is used exactly once, after which the original, error-generating
+ * handler is re-installed.  Thus, each event handler is used at most once.
+ *
+ * @param {object} wrapper
+ *        The wrapper on which the psuedo-handler is installed
+ * @param {object} obj
+ *        The real source of events
+ * @param {string} event
+ *        The name of the event
+ */
+function createOneShotEventWrapper(wrapper, obj, event) {
+  var onx = "on" + event;
+  var unexpected = unexpectedEvent(wrapper, event);
+  wrapper[onx] = unexpected;
+  obj[onx] = e => {
+    info(wrapper + ': "on' + event + '" event fired');
+    e.wrapper = wrapper;
+    wrapper[onx](e);
+    wrapper[onx] = unexpected;
+  };
+}
+
+/**
+ * Returns a promise that resolves when `target` has raised an event with the
+ * given name the given number of times. Cancel the returned promise by passing
+ * in a `cancel` promise and resolving it.
+ *
+ * @param {object} target
+ *        The target on which the event should occur.
+ * @param {string} name
+ *        The name of the event that should occur.
+ * @param {integer} count
+ *        Optional number of times the event should be raised before resolving.
+ * @param {promise} cancel
+ *        Optional promise that on resolving rejects the returned promise,
+ *        so we can avoid logging results after a test has finished.
+ * @returns {promise} A promise that resolves to the last of the seen events.
+ */
+function haveEvents(target, name, count, cancel) {
+  var listener;
+  var counter = count || 1;
+  return Promise.race([
+    (cancel || new Promise(() => {})).then(e => Promise.reject(e)),
+    new Promise(resolve =>
+      target.addEventListener(
+        name,
+        (listener = e => --counter < 1 && resolve(e))
+      )
+    ),
+  ]).then(e => (target.removeEventListener(name, listener), e));
+}
+
+/**
+ * Returns a promise that resolves when `target` has raised an event with the
+ * given name. Cancel the returned promise by passing in a `cancel` promise and
+ * resolving it.
+ *
+ * @param {object} target
+ *        The target on which the event should occur.
+ * @param {string} name
+ *        The name of the event that should occur.
+ * @param {promise} cancel
+ *        Optional promise that on resolving rejects the returned promise,
+ *        so we can avoid logging results after a test has finished.
+ * @returns {promise} A promise that resolves to the seen event.
+ */
+function haveEvent(target, name, cancel) {
+  return haveEvents(target, name, 1, cancel);
+}
+
+/**
+ * Returns a promise that resolves if the target has not seen the given event
+ * after one crank (or until the given timeoutPromise resolves) of the event
+ * loop.
+ *
+ * @param {object} target
+ *        The target on which the event should not occur.
+ * @param {string} name
+ *        The name of the event that should not occur.
+ * @param {promise} timeoutPromise
+ *        Optional promise defining how long we should wait before resolving.
+ * @returns {promise} A promise that is rejected if we see the given event, or
+ *                    resolves after a timeout otherwise.
+ */
+function haveNoEvent(target, name, timeoutPromise) {
+  return haveEvent(target, name, timeoutPromise || wait(0)).then(
+    () => Promise.reject(new Error("Too many " + name + " events")),
+    () => {}
+  );
+}
+
+/**
+ * Returns a promise that resolves after the target has seen the given number
+ * of events but no such event in a following crank of the event loop.
+ *
+ * @param {object} target
+ *        The target on which the events should occur.
+ * @param {string} name
+ *        The name of the event that should occur.
+ * @param {integer} count
+ *        Optional number of times the event should be raised before resolving.
+ * @param {promise} cancel
+ *        Optional promise that on resolving rejects the returned promise,
+ *        so we can avoid logging results after a test has finished.
+ * @returns {promise} A promise that resolves to the last of the seen events.
+ */
+function haveEventsButNoMore(target, name, count, cancel) {
+  return haveEvents(target, name, count, cancel).then(e =>
+    haveNoEvent(target, name).then(() => e)
+  );
+}
+
+/*
+ * Resolves the returned promise with an object with usage and reportCount
+ * properties.  `usage` is in the same units as reported by the reporter for
+ * `path`.
+ */
+const collectMemoryUsage = async path => {
+  const MemoryReporterManager = Cc[
+    "@mozilla.org/memory-reporter-manager;1"
+  ].getService(Ci.nsIMemoryReporterManager);
+
+  let usage = 0;
+  let reportCount = 0;
+  await new Promise(resolve =>
+    MemoryReporterManager.getReports(
+      (aProcess, aPath, aKind, aUnits, aAmount, aDesc) => {
+        if (aPath != path) {
+          return;
+        }
+        ++reportCount;
+        usage += aAmount;
+      },
+      null,
+      resolve,
+      null,
+      /* anonymized = */ false
+    )
+  );
+  return { usage, reportCount };
+};
+
+// Some DNS helper functions
+const dnsLookup = async hostname => {
+  // Convenience API for various networking related stuff. _Almost_ convenient
+  // enough.
+  const neckoDashboard = SpecialPowers.Cc[
+    "@mozilla.org/network/dashboard;1"
+  ].getService(Ci.nsIDashboard);
+
+  const results = await new Promise(r => {
+    neckoDashboard.requestDNSLookup(hostname, results => {
+      r(SpecialPowers.wrap(results));
+    });
+  });
+
+  // |address| is an array-like dictionary (ie; keys are all integers).
+  // We convert to an array to make it less unwieldy.
+  const addresses = [...results.address];
+  info(`DNS results for ${hostname}: ${JSON.stringify(addresses)}`);
+  return addresses;
+};
+
+const dnsLookupV4 = async hostname => {
+  const addresses = await dnsLookup(hostname);
+  return addresses.filter(address => !address.includes(":"));
+};
+
+const dnsLookupV6 = async hostname => {
+  const addresses = await dnsLookup(hostname);
+  return addresses.filter(address => address.includes(":"));
+};
+
+const getTurnHostname = turnUrl => {
+  const urlNoParams = turnUrl.split("?")[0];
+  // Strip off scheme
+  const hostAndMaybePort = urlNoParams.split(":", 2)[1];
+  if (hostAndMaybePort[0] == "[") {
+    // IPV6 literal, strip out '[', and split at closing ']'
+    return hostAndMaybePort.substring(1).split("]")[0];
+  }
+  return hostAndMaybePort.split(":")[0];
+};
+
+// Yo dawg I heard you like yo dawg I heard you like Proxies
+// Example: let value = await GleanTest.category.metric.testGetValue();
+// For labeled metrics:
+//    let value = await GleanTest.category.metric["label"].testGetValue();
+// Please don't try to use the string "testGetValue" as a label.
+const GleanTest = new Proxy(
+  {},
+  {
+    get(target, categoryName, receiver) {
+      return new Proxy(
+        {},
+        {
+          get(target, metricName, receiver) {
+            return new Proxy(
+              {
+                async testGetValue() {
+                  return SpecialPowers.spawnChrome(
+                    [categoryName, metricName],
+                    async (categoryName, metricName) => {
+                      await Services.fog.testFlushAllChildren();
+                      const window = this.browsingContext.topChromeWindow;
+                      return window.Glean[categoryName][
+                        metricName
+                      ].testGetValue();
+                    }
+                  );
+                },
+              },
+              {
+                get(target, prop, receiver) {
+                  // The only prop that will be there is testGetValue, but we
+                  // might add more later.
+                  if (prop in target) {
+                    return target[prop];
+                  }
+
+                  // |prop| must be a label?
+                  const label = prop;
+                  return {
+                    async testGetValue() {
+                      return SpecialPowers.spawnChrome(
+                        [categoryName, metricName, label],
+                        async (categoryName, metricName, label) => {
+                          await Services.fog.testFlushAllChildren();
+                          const window = this.browsingContext.topChromeWindow;
+                          return window.Glean[categoryName][metricName][
+                            label
+                          ].testGetValue();
+                        }
+                      );
+                    },
+                  };
+                },
+              }
+            );
+          },
+        }
+      );
+    },
+  }
+);
+
+/**
+ * This class executes a series of functions in a continuous sequence.
+ * Promise-bearing functions are executed after the previous promise completes.
+ *
+ * @constructor
+ * @param {object} framework
+ *        A back reference to the framework which makes use of the class. It is
+ *        passed to each command callback.
+ * @param {function[]} commandList
+ *        Commands to set during initialization
+ */
+function CommandChain(framework, commandList) {
+  this._framework = framework;
+  this.commands = commandList || [];
+}
+
+CommandChain.prototype = {
+  /**
+   * Start the command chain.  This returns a promise that always resolves
+   * cleanly (this catches errors and fails the test case).
+   */
+  execute() {
+    return this.commands
+      .reduce((prev, next, i) => {
+        if (typeof next !== "function" || !next.name) {
+          throw new Error("registered non-function" + next);
+        }
+
+        return prev.then(() => {
+          info("Run step " + (i + 1) + ": " + next.name);
+          return Promise.race([next(this._framework), rejectOnUnexpectedEvent]);
+        });
+      }, Promise.resolve())
+      .catch(e =>
+        ok(
+          false,
+          "Error in test execution: " +
+            e +
+            (typeof e.stack === "string"
+              ? " " + e.stack.split("\n").join(" ... ")
+              : "")
+        )
+      );
+  },
+
+  /**
+   * Add new commands to the end of the chain
+   */
+  append(commands) {
+    this.commands = this.commands.concat(commands);
+  },
+
+  /**
+   * Returns the index of the specified command in the chain.
+   * @param {occurrence} Optional param specifying which occurrence to match,
+   * with 0 representing the first occurrence.
+   */
+  indexOf(functionOrName, occurrence) {
+    occurrence = occurrence || 0;
+    return this.commands.findIndex(func => {
+      if (typeof functionOrName === "string") {
+        if (func.name !== functionOrName) {
+          return false;
+        }
+      } else if (func !== functionOrName) {
+        return false;
+      }
+      if (occurrence) {
+        --occurrence;
+        return false;
+      }
+      return true;
+    });
+  },
+
+  mustHaveIndexOf(functionOrName, occurrence) {
+    var index = this.indexOf(functionOrName, occurrence);
+    if (index == -1) {
+      throw new Error("Unknown test: " + functionOrName);
+    }
+    return index;
+  },
+
+  /**
+   * Inserts the new commands after the specified command.
+   */
+  insertAfter(functionOrName, commands, all, occurrence) {
+    this._insertHelper(functionOrName, commands, 1, all, occurrence);
+  },
+
+  /**
+   * Inserts the new commands after every occurrence of the specified command
+   */
+  insertAfterEach(functionOrName, commands) {
+    this._insertHelper(functionOrName, commands, 1, true);
+  },
+
+  /**
+   * Inserts the new commands before the specified command.
+   */
+  insertBefore(functionOrName, commands, all, occurrence) {
+    this._insertHelper(functionOrName, commands, 0, all, occurrence);
+  },
+
+  _insertHelper(functionOrName, commands, delta, all, occurrence) {
+    occurrence = occurrence || 0;
+    for (
+      var index = this.mustHaveIndexOf(functionOrName, occurrence);
+      index !== -1;
+      index = this.indexOf(functionOrName, ++occurrence)
+    ) {
+      this.commands = [].concat(
+        this.commands.slice(0, index + delta),
+        commands,
+        this.commands.slice(index + delta)
+      );
+      if (!all) {
+        break;
+      }
+    }
+  },
+
+  /**
+   * Removes the specified command, returns what was removed.
+   */
+  remove(functionOrName, occurrence) {
+    return this.commands.splice(
+      this.mustHaveIndexOf(functionOrName, occurrence),
+      1
+    );
+  },
+
+  /**
+   * Removes all commands after the specified one, returns what was removed.
+   */
+  removeAfter(functionOrName, occurrence) {
+    return this.commands.splice(
+      this.mustHaveIndexOf(functionOrName, occurrence) + 1
+    );
+  },
+
+  /**
+   * Removes all commands before the specified one, returns what was removed.
+   */
+  removeBefore(functionOrName, occurrence) {
+    return this.commands.splice(
+      0,
+      this.mustHaveIndexOf(functionOrName, occurrence)
+    );
+  },
+
+  /**
+   * Replaces a single command, returns what was removed.
+   */
+  replace(functionOrName, commands) {
+    this.insertBefore(functionOrName, commands);
+    return this.remove(functionOrName);
+  },
+
+  /**
+   * Replaces all commands after the specified one, returns what was removed.
+   */
+  replaceAfter(functionOrName, commands, occurrence) {
+    var oldCommands = this.removeAfter(functionOrName, occurrence);
+    this.append(commands);
+    return oldCommands;
+  },
+
+  /**
+   * Replaces all commands before the specified one, returns what was removed.
+   */
+  replaceBefore(functionOrName, commands) {
+    var oldCommands = this.removeBefore(functionOrName);
+    this.insertBefore(functionOrName, commands);
+    return oldCommands;
+  },
+
+  /**
+   * Remove all commands whose name match the specified regex.
+   */
+  filterOut(id_match) {
+    this.commands = this.commands.filter(c => !id_match.test(c.name));
+  },
+};
+
+function AudioStreamHelper() {
+  this._context = new AudioContext();
+}
+
+AudioStreamHelper.prototype = {
+  checkAudio(stream, analyser, fun) {
+    /*
+    analyser.enableDebugCanvas();
+    return analyser.waitForAnalysisSuccess(fun)
+      .then(() => analyser.disableDebugCanvas());
+    */
+    return analyser.waitForAnalysisSuccess(fun);
+  },
+
+  checkAudioFlowing(stream) {
+    var analyser = new AudioStreamAnalyser(this._context, stream);
+    var freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+    return this.checkAudio(stream, analyser, array => array[freq] > 200);
+  },
+
+  checkAudioNotFlowing(stream) {
+    var analyser = new AudioStreamAnalyser(this._context, stream);
+    var freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+    return this.checkAudio(stream, analyser, array => array[freq] < 50);
+  },
+};
+
+class VideoFrameEmitter {
+  constructor(color1, color2, width, height) {
+    if (!width) {
+      width = 50;
+    }
+    if (!height) {
+      height = width;
+    }
+    this._helper = new CaptureStreamTestHelper2D(width, height);
+    this._canvas = this._helper.createAndAppendElement(
+      "canvas",
+      "source_canvas"
+    );
+    this._canvas.width = width;
+    this._canvas.height = height;
+    this._color1 = color1 ? color1 : this._helper.green;
+    this._color2 = color2 ? color2 : this._helper.red;
+    // Make sure this is initted
+    this._helper.drawColor(this._canvas, this._color1);
+    this._stream = this._canvas.captureStream();
+    this._started = false;
+  }
+
+  stream() {
+    return this._stream;
+  }
+
+  helper() {
+    return this._helper;
+  }
+
+  colors(color1, color2) {
+    this._color1 = color1 ? color1 : this._helper.green;
+    this._color2 = color2 ? color2 : this._helper.red;
+    try {
+      this._helper.drawColor(this._canvas, this._color1);
+    } catch (e) {
+      // ignore; stream might have shut down
+    }
+  }
+
+  size(width, height) {
+    this._canvas.width = width;
+    this._canvas.height = height;
+  }
+
+  start() {
+    if (this._started) {
+      info("*** emitter already started");
+      return;
+    }
+
+    let i = 0;
+    this._started = true;
+    this._intervalId = setInterval(() => {
+      try {
+        this._helper.drawColor(this._canvas, i ? this._color1 : this._color2);
+        i = 1 - i;
+      } catch (e) {
+        // ignore; stream might have shut down, and we don't bother clearing
+        // the setInterval.
+      }
+    }, 500);
+  }
+
+  stop() {
+    if (this._started) {
+      clearInterval(this._intervalId);
+      this._started = false;
+    }
+  }
+}
+
+class VideoStreamHelper {
+  constructor() {
+    this._helper = new CaptureStreamTestHelper2D(50, 50);
+  }
+
+  async checkHasFrame(video, { offsetX, offsetY, threshold } = {}) {
+    const h = this._helper;
+    await h.waitForPixel(
+      video,
+      px => {
+        let result = h.isOpaquePixelNot(px, h.black, threshold);
+        info(
+          "Checking that we have a frame, got [" +
+            Array.from(px) +
+            "]. Ref=[" +
+            Array.from(h.black.data) +
+            "]. Threshold=" +
+            threshold +
+            ". Pass=" +
+            result
+        );
+        return result;
+      },
+      { offsetX, offsetY }
+    );
+  }
+
+  async checkVideoPlaying(
+    video,
+    { offsetX = 10, offsetY = 10, threshold = 16 } = {}
+  ) {
+    const h = this._helper;
+    await this.checkHasFrame(video, { offsetX, offsetY, threshold });
+    let startPixel = {
+      data: h.getPixel(video, offsetX, offsetY),
+      name: "startcolor",
+    };
+    await h.waitForPixel(
+      video,
+      px => {
+        let result = h.isPixelNot(px, startPixel, threshold);
+        info(
+          "Checking playing, [" +
+            Array.from(px) +
+            "] vs [" +
+            Array.from(startPixel.data) +
+            "]. Threshold=" +
+            threshold +
+            " Pass=" +
+            result
+        );
+        return result;
+      },
+      { offsetX, offsetY }
+    );
+  }
+
+  async checkVideoPaused(
+    video,
+    { offsetX = 10, offsetY = 10, threshold = 16, time = 5000 } = {}
+  ) {
+    const h = this._helper;
+    await this.checkHasFrame(video, { offsetX, offsetY, threshold });
+    let startPixel = {
+      data: h.getPixel(video, offsetX, offsetY),
+      name: "startcolor",
+    };
+    try {
+      await h.waitForPixel(
+        video,
+        px => {
+          let result = h.isOpaquePixelNot(px, startPixel, threshold);
+          info(
+            "Checking paused, [" +
+              Array.from(px) +
+              "] vs [" +
+              Array.from(startPixel.data) +
+              "]. Threshold=" +
+              threshold +
+              " Pass=" +
+              result
+          );
+          return result;
+        },
+        { offsetX, offsetY, cancel: wait(time, "timeout") }
+      );
+      ok(false, "Frame changed within " + time / 1000 + " seconds");
+    } catch (e) {
+      is(
+        e,
+        "timeout",
+        "Frame shouldn't change for " + time / 1000 + " seconds"
+      );
+    }
+  }
+}
+
+(function () {
+  var el = document.createElement("link");
+  el.rel = "stylesheet";
+  el.type = "text/css";
+  el.href = "/tests/SimpleTest/test.css";
+  document.head.appendChild(el);
+})();
diff --git a/dom/media/webrtc/tests/mochitests/helpers_from_wpt/sdp.js b/dom/media/webrtc/tests/mochitests/helpers_from_wpt/sdp.js
new file mode 100644
index 0000000000..6460f64a44
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/helpers_from_wpt/sdp.js
@@ -0,0 +1,889 @@
+/* eslint-env node */
+"use strict";
+
+// SDP helpers.
+var SDPUtils = {};
+
+// Generate an alphanumeric identifier for cname or mids.
+// TODO: use UUIDs instead? https://gist.github.com/jed/982883
+SDPUtils.generateIdentifier = function() {
+  return Math.random()
+    .toString(36)
+    .substr(2, 10);
+};
+
+// The RTCP CNAME used by all peerconnections from the same JS.
+SDPUtils.localCName = SDPUtils.generateIdentifier();
+
+// Splits SDP into lines, dealing with both CRLF and LF.
+SDPUtils.splitLines = function(blob) {
+  return blob
+    .trim()
+    .split("\n")
+    .map(function(line) {
+      return line.trim();
+    });
+};
+// Splits SDP into sessionpart and mediasections. Ensures CRLF.
+SDPUtils.splitSections = function(blob) {
+  var parts = blob.split("\nm=");
+  return parts.map(function(part, index) {
+    return (index > 0 ? "m=" + part : part).trim() + "\r\n";
+  });
+};
+
+// returns the session description.
+SDPUtils.getDescription = function(blob) {
+  var sections = SDPUtils.splitSections(blob);
+  return sections && sections[0];
+};
+
+// returns the individual media sections.
+SDPUtils.getMediaSections = function(blob) {
+  var sections = SDPUtils.splitSections(blob);
+  sections.shift();
+  return sections;
+};
+
+// Returns lines that start with a certain prefix.
+SDPUtils.matchPrefix = function(blob, prefix) {
+  return SDPUtils.splitLines(blob).filter(function(line) {
+    return line.indexOf(prefix) === 0;
+  });
+};
+
+// Parses an ICE candidate line. Sample input:
+// candidate:702786350 2 udp 41819902 8.8.8.8 60769 typ relay raddr 8.8.8.8
+// rport 55996"
+SDPUtils.parseCandidate = function(line) {
+  var parts;
+  // Parse both variants.
+  if (line.indexOf("a=candidate:") === 0) {
+    parts = line.substring(12).split(" ");
+  } else {
+    parts = line.substring(10).split(" ");
+  }
+
+  var candidate = {
+    foundation: parts[0],
+    component: parseInt(parts[1], 10),
+    protocol: parts[2].toLowerCase(),
+    priority: parseInt(parts[3], 10),
+    ip: parts[4],
+    address: parts[4], // address is an alias for ip.
+    port: parseInt(parts[5], 10),
+    // skip parts[6] == 'typ'
+    type: parts[7],
+  };
+
+  for (var i = 8; i < parts.length; i += 2) {
+    switch (parts[i]) {
+      case "raddr":
+        candidate.relatedAddress = parts[i + 1];
+        break;
+      case "rport":
+        candidate.relatedPort = parseInt(parts[i + 1], 10);
+        break;
+      case "tcptype":
+        candidate.tcpType = parts[i + 1];
+        break;
+      case "ufrag":
+        candidate.ufrag = parts[i + 1]; // for backward compability.
+        candidate.usernameFragment = parts[i + 1];
+        break;
+      default:
+        // extension handling, in particular ufrag
+        candidate[parts[i]] = parts[i + 1];
+        break;
+    }
+  }
+  return candidate;
+};
+
+// Translates a candidate object into SDP candidate attribute.
+SDPUtils.writeCandidate = function(candidate) {
+  var sdp = [];
+  sdp.push(candidate.foundation);
+  sdp.push(candidate.component);
+  sdp.push(candidate.protocol.toUpperCase());
+  sdp.push(candidate.priority);
+  sdp.push(candidate.address || candidate.ip);
+  sdp.push(candidate.port);
+
+  var type = candidate.type;
+  sdp.push("typ");
+  sdp.push(type);
+  if (type !== "host" && candidate.relatedAddress && candidate.relatedPort) {
+    sdp.push("raddr");
+    sdp.push(candidate.relatedAddress);
+    sdp.push("rport");
+    sdp.push(candidate.relatedPort);
+  }
+  if (candidate.tcpType && candidate.protocol.toLowerCase() === "tcp") {
+    sdp.push("tcptype");
+    sdp.push(candidate.tcpType);
+  }
+  if (candidate.usernameFragment || candidate.ufrag) {
+    sdp.push("ufrag");
+    sdp.push(candidate.usernameFragment || candidate.ufrag);
+  }
+  return "candidate:" + sdp.join(" ");
+};
+
+// Parses an ice-options line, returns an array of option tags.
+// a=ice-options:foo bar
+SDPUtils.parseIceOptions = function(line) {
+  return line.substr(14).split(" ");
+};
+
+// Parses an rtpmap line, returns RTCRtpCoddecParameters. Sample input:
+// a=rtpmap:111 opus/48000/2
+SDPUtils.parseRtpMap = function(line) {
+  var parts = line.substr(9).split(" ");
+  var parsed = {
+    payloadType: parseInt(parts.shift(), 10), // was: id
+  };
+
+  parts = parts[0].split("/");
+
+  parsed.name = parts[0];
+  parsed.clockRate = parseInt(parts[1], 10); // was: clockrate
+  parsed.channels = parts.length === 3 ? parseInt(parts[2], 10) : 1;
+  // legacy alias, got renamed back to channels in ORTC.
+  parsed.numChannels = parsed.channels;
+  return parsed;
+};
+
+// Generate an a=rtpmap line from RTCRtpCodecCapability or
+// RTCRtpCodecParameters.
+SDPUtils.writeRtpMap = function(codec) {
+  var pt = codec.payloadType;
+  if (codec.preferredPayloadType !== undefined) {
+    pt = codec.preferredPayloadType;
+  }
+  var channels = codec.channels || codec.numChannels || 1;
+  return (
+    "a=rtpmap:" +
+    pt +
+    " " +
+    codec.name +
+    "/" +
+    codec.clockRate +
+    (channels !== 1 ? "/" + channels : "") +
+    "\r\n"
+  );
+};
+
+// Parses an a=extmap line (headerextension from RFC 5285). Sample input:
+// a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
+// a=extmap:2/sendonly urn:ietf:params:rtp-hdrext:toffset
+SDPUtils.parseExtmap = function(line) {
+  var parts = line.substr(9).split(" ");
+  return {
+    id: parseInt(parts[0], 10),
+    direction: parts[0].indexOf("/") > 0 ? parts[0].split("/")[1] : "sendrecv",
+    uri: parts[1],
+  };
+};
+
+// Generates a=extmap line from RTCRtpHeaderExtensionParameters or
+// RTCRtpHeaderExtension.
+SDPUtils.writeExtmap = function(headerExtension) {
+  return (
+    "a=extmap:" +
+    (headerExtension.id || headerExtension.preferredId) +
+    (headerExtension.direction && headerExtension.direction !== "sendrecv"
+      ? "/" + headerExtension.direction
+      : "") +
+    " " +
+    headerExtension.uri +
+    "\r\n"
+  );
+};
+
+// Parses an ftmp line, returns dictionary. Sample input:
+// a=fmtp:96 vbr=on;cng=on
+// Also deals with vbr=on; cng=on
+SDPUtils.parseFmtp = function(line) {
+  var parsed = {};
+  var kv;
+  var parts = line.substr(line.indexOf(" ") + 1).split(";");
+  for (var j = 0; j < parts.length; j++) {
+    kv = parts[j].trim().split("=");
+    parsed[kv[0].trim()] = kv[1];
+  }
+  return parsed;
+};
+
+// Generates an a=ftmp line from RTCRtpCodecCapability or RTCRtpCodecParameters.
+SDPUtils.writeFmtp = function(codec) {
+  var line = "";
+  var pt = codec.payloadType;
+  if (codec.preferredPayloadType !== undefined) {
+    pt = codec.preferredPayloadType;
+  }
+  if (codec.parameters && Object.keys(codec.parameters).length) {
+    var params = [];
+    Object.keys(codec.parameters).forEach(function(param) {
+      if (codec.parameters[param]) {
+        params.push(param + "=" + codec.parameters[param]);
+      } else {
+        params.push(param);
+      }
+    });
+    line += "a=fmtp:" + pt + " " + params.join(";") + "\r\n";
+  }
+  return line;
+};
+
+// Parses an rtcp-fb line, returns RTCPRtcpFeedback object. Sample input:
+// a=rtcp-fb:98 nack rpsi
+SDPUtils.parseRtcpFb = function(line) {
+  var parts = line.substr(line.indexOf(" ") + 1).split(" ");
+  return {
+    type: parts.shift(),
+    parameter: parts.join(" "),
+  };
+};
+// Generate a=rtcp-fb lines from RTCRtpCodecCapability or RTCRtpCodecParameters.
+SDPUtils.writeRtcpFb = function(codec) {
+  var lines = "";
+  var pt = codec.payloadType;
+  if (codec.preferredPayloadType !== undefined) {
+    pt = codec.preferredPayloadType;
+  }
+  if (codec.rtcpFeedback && codec.rtcpFeedback.length) {
+    // FIXME: special handling for trr-int?
+    codec.rtcpFeedback.forEach(function(fb) {
+      lines +=
+        "a=rtcp-fb:" +
+        pt +
+        " " +
+        fb.type +
+        (fb.parameter && fb.parameter.length ? " " + fb.parameter : "") +
+        "\r\n";
+    });
+  }
+  return lines;
+};
+
+// Parses an RFC 5576 ssrc media attribute. Sample input:
+// a=ssrc:3735928559 cname:something
+SDPUtils.parseSsrcMedia = function(line) {
+  var sp = line.indexOf(" ");
+  var parts = {
+    ssrc: parseInt(line.substr(7, sp - 7), 10),
+  };
+  var colon = line.indexOf(":", sp);
+  if (colon > -1) {
+    parts.attribute = line.substr(sp + 1, colon - sp - 1);
+    parts.value = line.substr(colon + 1);
+  } else {
+    parts.attribute = line.substr(sp + 1);
+  }
+  return parts;
+};
+
+SDPUtils.parseSsrcGroup = function(line) {
+  var parts = line.substr(13).split(" ");
+  return {
+    semantics: parts.shift(),
+    ssrcs: parts.map(function(ssrc) {
+      return parseInt(ssrc, 10);
+    }),
+  };
+};
+
+// Extracts the MID (RFC 5888) from a media section.
+// returns the MID or undefined if no mid line was found.
+SDPUtils.getMid = function(mediaSection) {
+  var mid = SDPUtils.matchPrefix(mediaSection, "a=mid:")[0];
+  if (mid) {
+    return mid.substr(6);
+  }
+};
+
+SDPUtils.parseFingerprint = function(line) {
+  var parts = line.substr(14).split(" ");
+  return {
+    algorithm: parts[0].toLowerCase(), // algorithm is case-sensitive in Edge.
+    value: parts[1],
+  };
+};
+
+// Extracts DTLS parameters from SDP media section or sessionpart.
+// FIXME: for consistency with other functions this should only
+//   get the fingerprint line as input. See also getIceParameters.
+SDPUtils.getDtlsParameters = function(mediaSection, sessionpart) {
+  var lines = SDPUtils.matchPrefix(
+    mediaSection + sessionpart,
+    "a=fingerprint:"
+  );
+  // Note: a=setup line is ignored since we use the 'auto' role.
+  // Note2: 'algorithm' is not case sensitive except in Edge.
+  return {
+    role: "auto",
+    fingerprints: lines.map(SDPUtils.parseFingerprint),
+  };
+};
+
+// Serializes DTLS parameters to SDP.
+SDPUtils.writeDtlsParameters = function(params, setupType) {
+  var sdp = "a=setup:" + setupType + "\r\n";
+  params.fingerprints.forEach(function(fp) {
+    sdp += "a=fingerprint:" + fp.algorithm + " " + fp.value + "\r\n";
+  });
+  return sdp;
+};
+
+// Parses a=crypto lines into
+//   https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#dictionary-rtcsrtpsdesparameters-members
+SDPUtils.parseCryptoLine = function(line) {
+  var parts = line.substr(9).split(" ");
+  return {
+    tag: parseInt(parts[0], 10),
+    cryptoSuite: parts[1],
+    keyParams: parts[2],
+    sessionParams: parts.slice(3),
+  };
+};
+
+SDPUtils.writeCryptoLine = function(parameters) {
+  return (
+    "a=crypto:" +
+    parameters.tag +
+    " " +
+    parameters.cryptoSuite +
+    " " +
+    (typeof parameters.keyParams === "object"
+      ? SDPUtils.writeCryptoKeyParams(parameters.keyParams)
+      : parameters.keyParams) +
+    (parameters.sessionParams ? " " + parameters.sessionParams.join(" ") : "") +
+    "\r\n"
+  );
+};
+
+// Parses the crypto key parameters into
+//   https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#rtcsrtpkeyparam*
+SDPUtils.parseCryptoKeyParams = function(keyParams) {
+  if (keyParams.indexOf("inline:") !== 0) {
+    return null;
+  }
+  var parts = keyParams.substr(7).split("|");
+  return {
+    keyMethod: "inline",
+    keySalt: parts[0],
+    lifeTime: parts[1],
+    mkiValue: parts[2] ? parts[2].split(":")[0] : undefined,
+    mkiLength: parts[2] ? parts[2].split(":")[1] : undefined,
+  };
+};
+
+SDPUtils.writeCryptoKeyParams = function(keyParams) {
+  return (
+    keyParams.keyMethod +
+    ":" +
+    keyParams.keySalt +
+    (keyParams.lifeTime ? "|" + keyParams.lifeTime : "") +
+    (keyParams.mkiValue && keyParams.mkiLength
+      ? "|" + keyParams.mkiValue + ":" + keyParams.mkiLength
+      : "")
+  );
+};
+
+// Extracts all SDES paramters.
+SDPUtils.getCryptoParameters = function(mediaSection, sessionpart) {
+  var lines = SDPUtils.matchPrefix(mediaSection + sessionpart, "a=crypto:");
+  return lines.map(SDPUtils.parseCryptoLine);
+};
+
+// Parses ICE information from SDP media section or sessionpart.
+// FIXME: for consistency with other functions this should only
+//   get the ice-ufrag and ice-pwd lines as input.
+SDPUtils.getIceParameters = function(mediaSection, sessionpart) {
+  var ufrag = SDPUtils.matchPrefix(
+    mediaSection + sessionpart,
+    "a=ice-ufrag:"
+  )[0];
+  var pwd = SDPUtils.matchPrefix(mediaSection + sessionpart, "a=ice-pwd:")[0];
+  if (!(ufrag && pwd)) {
+    return null;
+  }
+  return {
+    usernameFragment: ufrag.substr(12),
+    password: pwd.substr(10),
+  };
+};
+
+// Serializes ICE parameters to SDP.
+SDPUtils.writeIceParameters = function(params) {
+  return (
+    "a=ice-ufrag:" +
+    params.usernameFragment +
+    "\r\n" +
+    "a=ice-pwd:" +
+    params.password +
+    "\r\n"
+  );
+};
+
+// Parses the SDP media section and returns RTCRtpParameters.
+SDPUtils.parseRtpParameters = function(mediaSection) {
+  var description = {
+    codecs: [],
+    headerExtensions: [],
+    fecMechanisms: [],
+    rtcp: [],
+  };
+  var lines = SDPUtils.splitLines(mediaSection);
+  var mline = lines[0].split(" ");
+  for (var i = 3; i < mline.length; i++) {
+    // find all codecs from mline[3..]
+    var pt = mline[i];
+    var rtpmapline = SDPUtils.matchPrefix(
+      mediaSection,
+      "a=rtpmap:" + pt + " "
+    )[0];
+    if (rtpmapline) {
+      var codec = SDPUtils.parseRtpMap(rtpmapline);
+      var fmtps = SDPUtils.matchPrefix(mediaSection, "a=fmtp:" + pt + " ");
+      // Only the first a=fmtp:<pt> is considered.
+      codec.parameters = fmtps.length ? SDPUtils.parseFmtp(fmtps[0]) : {};
+      codec.rtcpFeedback = SDPUtils.matchPrefix(
+        mediaSection,
+        "a=rtcp-fb:" + pt + " "
+      ).map(SDPUtils.parseRtcpFb);
+      description.codecs.push(codec);
+      // parse FEC mechanisms from rtpmap lines.
+      switch (codec.name.toUpperCase()) {
+        case "RED":
+        case "ULPFEC":
+          description.fecMechanisms.push(codec.name.toUpperCase());
+          break;
+        default:
+          // only RED and ULPFEC are recognized as FEC mechanisms.
+          break;
+      }
+    }
+  }
+  SDPUtils.matchPrefix(mediaSection, "a=extmap:").forEach(function(line) {
+    description.headerExtensions.push(SDPUtils.parseExtmap(line));
+  });
+  // FIXME: parse rtcp.
+  return description;
+};
+
+// Generates parts of the SDP media section describing the capabilities /
+// parameters.
+SDPUtils.writeRtpDescription = function(kind, caps) {
+  var sdp = "";
+
+  // Build the mline.
+  sdp += "m=" + kind + " ";
+  sdp += caps.codecs.length > 0 ? "9" : "0"; // reject if no codecs.
+  sdp += " UDP/TLS/RTP/SAVPF ";
+  sdp +=
+    caps.codecs
+      .map(function(codec) {
+        if (codec.preferredPayloadType !== undefined) {
+          return codec.preferredPayloadType;
+        }
+        return codec.payloadType;
+      })
+      .join(" ") + "\r\n";
+
+  sdp += "c=IN IP4 0.0.0.0\r\n";
+  sdp += "a=rtcp:9 IN IP4 0.0.0.0\r\n";
+
+  // Add a=rtpmap lines for each codec. Also fmtp and rtcp-fb.
+  caps.codecs.forEach(function(codec) {
+    sdp += SDPUtils.writeRtpMap(codec);
+    sdp += SDPUtils.writeFmtp(codec);
+    sdp += SDPUtils.writeRtcpFb(codec);
+  });
+  var maxptime = 0;
+  caps.codecs.forEach(function(codec) {
+    if (codec.maxptime > maxptime) {
+      maxptime = codec.maxptime;
+    }
+  });
+  if (maxptime > 0) {
+    sdp += "a=maxptime:" + maxptime + "\r\n";
+  }
+  sdp += "a=rtcp-mux\r\n";
+
+  if (caps.headerExtensions) {
+    caps.headerExtensions.forEach(function(extension) {
+      sdp += SDPUtils.writeExtmap(extension);
+    });
+  }
+  // FIXME: write fecMechanisms.
+  return sdp;
+};
+
+// Parses the SDP media section and returns an array of
+// RTCRtpEncodingParameters.
+SDPUtils.parseRtpEncodingParameters = function(mediaSection) {
+  var encodingParameters = [];
+  var description = SDPUtils.parseRtpParameters(mediaSection);
+  var hasRed = description.fecMechanisms.indexOf("RED") !== -1;
+  var hasUlpfec = description.fecMechanisms.indexOf("ULPFEC") !== -1;
+
+  // filter a=ssrc:... cname:, ignore PlanB-msid
+  var ssrcs = SDPUtils.matchPrefix(mediaSection, "a=ssrc:")
+    .map(function(line) {
+      return SDPUtils.parseSsrcMedia(line);
+    })
+    .filter(function(parts) {
+      return parts.attribute === "cname";
+    });
+  var primarySsrc = ssrcs.length > 0 && ssrcs[0].ssrc;
+  var secondarySsrc;
+
+  var flows = SDPUtils.matchPrefix(mediaSection, "a=ssrc-group:FID").map(
+    function(line) {
+      var parts = line.substr(17).split(" ");
+      return parts.map(function(part) {
+        return parseInt(part, 10);
+      });
+    }
+  );
+  if (flows.length > 0 && flows[0].length > 1 && flows[0][0] === primarySsrc) {
+    secondarySsrc = flows[0][1];
+  }
+
+  description.codecs.forEach(function(codec) {
+    if (codec.name.toUpperCase() === "RTX" && codec.parameters.apt) {
+      var encParam = {
+        ssrc: primarySsrc,
+        codecPayloadType: parseInt(codec.parameters.apt, 10),
+      };
+      if (primarySsrc && secondarySsrc) {
+        encParam.rtx = { ssrc: secondarySsrc };
+      }
+      encodingParameters.push(encParam);
+      if (hasRed) {
+        encParam = JSON.parse(JSON.stringify(encParam));
+        encParam.fec = {
+          ssrc: primarySsrc,
+          mechanism: hasUlpfec ? "red+ulpfec" : "red",
+        };
+        encodingParameters.push(encParam);
+      }
+    }
+  });
+  if (encodingParameters.length === 0 && primarySsrc) {
+    encodingParameters.push({
+      ssrc: primarySsrc,
+    });
+  }
+
+  // we support both b=AS and b=TIAS but interpret AS as TIAS.
+  var bandwidth = SDPUtils.matchPrefix(mediaSection, "b=");
+  if (bandwidth.length) {
+    if (bandwidth[0].indexOf("b=TIAS:") === 0) {
+      bandwidth = parseInt(bandwidth[0].substr(7), 10);
+    } else if (bandwidth[0].indexOf("b=AS:") === 0) {
+      // use formula from JSEP to convert b=AS to TIAS value.
+      bandwidth =
+        parseInt(bandwidth[0].substr(5), 10) * 1000 * 0.95 - 50 * 40 * 8;
+    } else {
+      bandwidth = undefined;
+    }
+    encodingParameters.forEach(function(params) {
+      params.maxBitrate = bandwidth;
+    });
+  }
+  return encodingParameters;
+};
+
+// parses http://draft.ortc.org/#rtcrtcpparameters*
+SDPUtils.parseRtcpParameters = function(mediaSection) {
+  var rtcpParameters = {};
+
+  // Gets the first SSRC. Note tha with RTX there might be multiple
+  // SSRCs.
+  var remoteSsrc = SDPUtils.matchPrefix(mediaSection, "a=ssrc:")
+    .map(function(line) {
+      return SDPUtils.parseSsrcMedia(line);
+    })
+    .filter(function(obj) {
+      return obj.attribute === "cname";
+    })[0];
+  if (remoteSsrc) {
+    rtcpParameters.cname = remoteSsrc.value;
+    rtcpParameters.ssrc = remoteSsrc.ssrc;
+  }
+
+  // Edge uses the compound attribute instead of reducedSize
+  // compound is !reducedSize
+  var rsize = SDPUtils.matchPrefix(mediaSection, "a=rtcp-rsize");
+  rtcpParameters.reducedSize = rsize.length > 0;
+  rtcpParameters.compound = rsize.length === 0;
+
+  // parses the rtcp-mux attrіbute.
+  // Note that Edge does not support unmuxed RTCP.
+  var mux = SDPUtils.matchPrefix(mediaSection, "a=rtcp-mux");
+  rtcpParameters.mux = mux.length > 0;
+
+  return rtcpParameters;
+};
+
+// parses either a=msid: or a=ssrc:... msid lines and returns
+// the id of the MediaStream and MediaStreamTrack.
+SDPUtils.parseMsid = function(mediaSection) {
+  var parts;
+  var spec = SDPUtils.matchPrefix(mediaSection, "a=msid:");
+  if (spec.length === 1) {
+    parts = spec[0].substr(7).split(" ");
+    return { stream: parts[0], track: parts[1] };
+  }
+  var planB = SDPUtils.matchPrefix(mediaSection, "a=ssrc:")
+    .map(function(line) {
+      return SDPUtils.parseSsrcMedia(line);
+    })
+    .filter(function(msidParts) {
+      return msidParts.attribute === "msid";
+    });
+  if (planB.length > 0) {
+    parts = planB[0].value.split(" ");
+    return { stream: parts[0], track: parts[1] };
+  }
+};
+
+// SCTP
+// parses draft-ietf-mmusic-sctp-sdp-26 first and falls back
+// to draft-ietf-mmusic-sctp-sdp-05
+SDPUtils.parseSctpDescription = function(mediaSection) {
+  var mline = SDPUtils.parseMLine(mediaSection);
+  var maxSizeLine = SDPUtils.matchPrefix(mediaSection, "a=max-message-size:");
+  var maxMessageSize;
+  if (maxSizeLine.length > 0) {
+    maxMessageSize = parseInt(maxSizeLine[0].substr(19), 10);
+  }
+  if (isNaN(maxMessageSize)) {
+    maxMessageSize = 65536;
+  }
+  var sctpPort = SDPUtils.matchPrefix(mediaSection, "a=sctp-port:");
+  if (sctpPort.length > 0) {
+    return {
+      port: parseInt(sctpPort[0].substr(12), 10),
+      protocol: mline.fmt,
+      maxMessageSize,
+    };
+  }
+  var sctpMapLines = SDPUtils.matchPrefix(mediaSection, "a=sctpmap:");
+  if (sctpMapLines.length > 0) {
+    var parts = SDPUtils.matchPrefix(mediaSection, "a=sctpmap:")[0]
+      .substr(10)
+      .split(" ");
+    return {
+      port: parseInt(parts[0], 10),
+      protocol: parts[1],
+      maxMessageSize,
+    };
+  }
+};
+
+// SCTP
+// outputs the draft-ietf-mmusic-sctp-sdp-26 version that all browsers
+// support by now receiving in this format, unless we originally parsed
+// as the draft-ietf-mmusic-sctp-sdp-05 format (indicated by the m-line
+// protocol of DTLS/SCTP -- without UDP/ or TCP/)
+SDPUtils.writeSctpDescription = function(media, sctp) {
+  var output = [];
+  if (media.protocol !== "DTLS/SCTP") {
+    output = [
+      "m=" + media.kind + " 9 " + media.protocol + " " + sctp.protocol + "\r\n",
+      "c=IN IP4 0.0.0.0\r\n",
+      "a=sctp-port:" + sctp.port + "\r\n",
+    ];
+  } else {
+    output = [
+      "m=" + media.kind + " 9 " + media.protocol + " " + sctp.port + "\r\n",
+      "c=IN IP4 0.0.0.0\r\n",
+      "a=sctpmap:" + sctp.port + " " + sctp.protocol + " 65535\r\n",
+    ];
+  }
+  if (sctp.maxMessageSize !== undefined) {
+    output.push("a=max-message-size:" + sctp.maxMessageSize + "\r\n");
+  }
+  return output.join("");
+};
+
+// Generate a session ID for SDP.
+// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-20#section-5.2.1
+// recommends using a cryptographically random +ve 64-bit value
+// but right now this should be acceptable and within the right range
+SDPUtils.generateSessionId = function() {
+  return Math.floor((Math.random() * 4294967296) + 1);
+};
+
+// Write boilder plate for start of SDP
+// sessId argument is optional - if not supplied it will
+// be generated randomly
+// sessVersion is optional and defaults to 2
+// sessUser is optional and defaults to 'thisisadapterortc'
+SDPUtils.writeSessionBoilerplate = function(sessId, sessVer, sessUser) {
+  var sessionId;
+  var version = sessVer !== undefined ? sessVer : 2;
+  if (sessId) {
+    sessionId = sessId;
+  } else {
+    sessionId = SDPUtils.generateSessionId();
+  }
+  var user = sessUser || "thisisadapterortc";
+  // FIXME: sess-id should be an NTP timestamp.
+  return (
+    "v=0\r\n" +
+    "o=" +
+    user +
+    " " +
+    sessionId +
+    " " +
+    version +
+    " IN IP4 127.0.0.1\r\n" +
+    "s=-\r\n" +
+    "t=0 0\r\n"
+  );
+};
+
+SDPUtils.writeMediaSection = function(transceiver, caps, type, stream) {
+  var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps);
+
+  // Map ICE parameters (ufrag, pwd) to SDP.
+  sdp += SDPUtils.writeIceParameters(
+    transceiver.iceGatherer.getLocalParameters()
+  );
+
+  // Map DTLS parameters to SDP.
+  sdp += SDPUtils.writeDtlsParameters(
+    transceiver.dtlsTransport.getLocalParameters(),
+    type === "offer" ? "actpass" : "active"
+  );
+
+  sdp += "a=mid:" + transceiver.mid + "\r\n";
+
+  if (transceiver.direction) {
+    sdp += "a=" + transceiver.direction + "\r\n";
+  } else if (transceiver.rtpSender && transceiver.rtpReceiver) {
+    sdp += "a=sendrecv\r\n";
+  } else if (transceiver.rtpSender) {
+    sdp += "a=sendonly\r\n";
+  } else if (transceiver.rtpReceiver) {
+    sdp += "a=recvonly\r\n";
+  } else {
+    sdp += "a=inactive\r\n";
+  }
+
+  if (transceiver.rtpSender) {
+    // spec.
+    var msid =
+      "msid:" + stream.id + " " + transceiver.rtpSender.track.id + "\r\n";
+    sdp += "a=" + msid;
+
+    // for Chrome.
+    sdp += "a=ssrc:" + transceiver.sendEncodingParameters[0].ssrc + " " + msid;
+    if (transceiver.sendEncodingParameters[0].rtx) {
+      sdp +=
+        "a=ssrc:" + transceiver.sendEncodingParameters[0].rtx.ssrc + " " + msid;
+      sdp +=
+        "a=ssrc-group:FID " +
+        transceiver.sendEncodingParameters[0].ssrc +
+        " " +
+        transceiver.sendEncodingParameters[0].rtx.ssrc +
+        "\r\n";
+    }
+  }
+  // FIXME: this should be written by writeRtpDescription.
+  sdp +=
+    "a=ssrc:" +
+    transceiver.sendEncodingParameters[0].ssrc +
+    " cname:" +
+    SDPUtils.localCName +
+    "\r\n";
+  if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) {
+    sdp +=
+      "a=ssrc:" +
+      transceiver.sendEncodingParameters[0].rtx.ssrc +
+      " cname:" +
+      SDPUtils.localCName +
+      "\r\n";
+  }
+  return sdp;
+};
+
+// Gets the direction from the mediaSection or the sessionpart.
+SDPUtils.getDirection = function(mediaSection, sessionpart) {
+  // Look for sendrecv, sendonly, recvonly, inactive, default to sendrecv.
+  var lines = SDPUtils.splitLines(mediaSection);
+  for (var i = 0; i < lines.length; i++) {
+    switch (lines[i]) {
+      case "a=sendrecv":
+      case "a=sendonly":
+      case "a=recvonly":
+      case "a=inactive":
+        return lines[i].substr(2);
+      default:
+      // FIXME: What should happen here?
+    }
+  }
+  if (sessionpart) {
+    return SDPUtils.getDirection(sessionpart);
+  }
+  return "sendrecv";
+};
+
+SDPUtils.getKind = function(mediaSection) {
+  var lines = SDPUtils.splitLines(mediaSection);
+  var mline = lines[0].split(" ");
+  return mline[0].substr(2);
+};
+
+SDPUtils.isRejected = function(mediaSection) {
+  return mediaSection.split(" ", 2)[1] === "0";
+};
+
+SDPUtils.parseMLine = function(mediaSection) {
+  var lines = SDPUtils.splitLines(mediaSection);
+  var parts = lines[0].substr(2).split(" ");
+  return {
+    kind: parts[0],
+    port: parseInt(parts[1], 10),
+    protocol: parts[2],
+    fmt: parts.slice(3).join(" "),
+  };
+};
+
+SDPUtils.parseOLine = function(mediaSection) {
+  var line = SDPUtils.matchPrefix(mediaSection, "o=")[0];
+  var parts = line.substr(2).split(" ");
+  return {
+    username: parts[0],
+    sessionId: parts[1],
+    sessionVersion: parseInt(parts[2], 10),
+    netType: parts[3],
+    addressType: parts[4],
+    address: parts[5],
+  };
+};
+
+// a very naive interpretation of a valid SDP.
+SDPUtils.isValidSDP = function(blob) {
+  if (typeof blob !== "string" || blob.length === 0) {
+    return false;
+  }
+  var lines = SDPUtils.splitLines(blob);
+  for (var i = 0; i < lines.length; i++) {
+    if (lines[i].length < 2 || lines[i].charAt(1) !== "=") {
+      return false;
+    }
+    // TODO: check the modifier a bit more.
+  }
+  return true;
+};
+
+// Expose public methods.
+if (typeof module === "object") {
+  module.exports = SDPUtils;
+}
diff --git a/dom/media/webrtc/tests/mochitests/iceTestUtils.js b/dom/media/webrtc/tests/mochitests/iceTestUtils.js
new file mode 100644
index 0000000000..d4d1f5c4b4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/iceTestUtils.js
@@ -0,0 +1,302 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+"use strict";
+
+// This is mostly so test_peerConnection_gatherWithStun300.html and
+// test_peerConnection_gatherWithStun300IPv6 can share this code. I would have
+// put the ipv6 test code in the same file, but our ipv6 tester support is
+// inconsistent enough that we need to be able to track the ipv6 test
+// separately.
+
+async function findStatsRelayCandidates(pc, protocol) {
+  const stats = await pc.getStats();
+  return [...stats.values()].filter(
+    v =>
+      v.type == "local-candidate" &&
+      v.candidateType == "relay" &&
+      v.relayProtocol == protocol
+  );
+}
+
+// Trickles candidates if pcDst is set, and resolves the candidate list
+async function trickleIce(pc, pcDst) {
+  const candidates = [],
+    addCandidatePromises = [];
+  while (true) {
+    const { candidate } = await new Promise(r =>
+      pc.addEventListener("icecandidate", r, { once: true })
+    );
+    if (!candidate) {
+      break;
+    }
+    candidates.push(candidate);
+    if (pcDst) {
+      addCandidatePromises.push(pcDst.addIceCandidate(candidate));
+    }
+  }
+  await Promise.all(addCandidatePromises);
+  return candidates;
+}
+
+async function gather(pc) {
+  if (pc.signalingState == "stable") {
+    await pc.setLocalDescription(
+      await pc.createOffer({ offerToReceiveAudio: true })
+    );
+  } else if (pc.signalingState == "have-remote-offer") {
+    await pc.setLocalDescription();
+  }
+
+  return trickleIce(pc);
+}
+
+async function gatherWithTimeout(pc, timeout, context) {
+  const throwOnTimeout = async () => {
+    await wait(timeout);
+    throw new Error(
+      `Gathering did not complete within ${timeout} ms with ${context}`
+    );
+  };
+
+  return Promise.race([gather(pc), throwOnTimeout()]);
+}
+
+async function iceConnected(pc) {
+  return new Promise((resolve, reject) => {
+    pc.addEventListener("iceconnectionstatechange", () => {
+      if (["connected", "completed"].includes(pc.iceConnectionState)) {
+        resolve();
+      } else if (pc.iceConnectionState == "failed") {
+        reject(new Error(`ICE failed`));
+      }
+    });
+  });
+}
+
+// Set up trickle, but does not wait for it to complete. Can be used by itself
+// in cases where we do not expect any new candidates, but want to still set up
+// the signal handling in case new candidates _do_ show up.
+async function connectNoTrickleWait(offerer, answerer, timeout, context) {
+  return connect(offerer, answerer, timeout, context, true);
+}
+
+async function connect(
+  offerer,
+  answerer,
+  timeout,
+  context,
+  noTrickleWait = false
+) {
+  const trickle1 = trickleIce(offerer, answerer);
+  const trickle2 = trickleIce(answerer, offerer);
+  try {
+    const offer = await offerer.createOffer({ offerToReceiveAudio: true });
+    await offerer.setLocalDescription(offer);
+    await answerer.setRemoteDescription(offer);
+    const answer = await answerer.createAnswer();
+    await Promise.all([
+      offerer.setRemoteDescription(answer),
+      answerer.setLocalDescription(answer),
+    ]);
+
+    const throwOnTimeout = async () => {
+      if (timeout) {
+        await wait(timeout);
+        throw new Error(
+          `ICE did not complete within ${timeout} ms with ${context}`
+        );
+      }
+    };
+
+    await Promise.race([
+      Promise.all([iceConnected(offerer), iceConnected(answerer)]),
+      throwOnTimeout(timeout, context),
+    ]);
+  } finally {
+    if (!noTrickleWait) {
+      // TODO(bug 1751509): For now, we need to let gathering finish before we
+      // proceed, because there are races in ICE restart wrt gathering state.
+      await Promise.all([trickle1, trickle2]);
+    }
+  }
+}
+
+function isV6HostCandidate(candidate) {
+  const fields = candidate.candidate.split(" ");
+  const type = fields[7];
+  const ipAddress = fields[4];
+  return type == "host" && ipAddress.includes(":");
+}
+
+async function ipv6Supported() {
+  const pc = new RTCPeerConnection();
+  const candidates = await gatherWithTimeout(pc, 8000);
+  info(`baseline candidates: ${JSON.stringify(candidates)}`);
+  pc.close();
+  return candidates.some(isV6HostCandidate);
+}
+
+function makeContextString(iceServers) {
+  const currentRedirectAddress = SpecialPowers.getCharPref(
+    "media.peerconnection.nat_simulator.redirect_address",
+    ""
+  );
+  const currentRedirectTargets = SpecialPowers.getCharPref(
+    "media.peerconnection.nat_simulator.redirect_targets",
+    ""
+  );
+  return `redirect rule: ${currentRedirectAddress}=>${currentRedirectTargets} iceServers: ${JSON.stringify(
+    iceServers
+  )}`;
+}
+
+async function checkSrflx(iceServers) {
+  const context = makeContextString(iceServers);
+  info(`checkSrflx ${context}`);
+  const pc = new RTCPeerConnection({
+    iceServers,
+    bundlePolicy: "max-bundle", // Avoids extra candidates
+  });
+  const candidates = await gatherWithTimeout(pc, 8000, context);
+  const srflxCandidates = candidates.filter(c => c.candidate.includes("srflx"));
+  info(`candidates: ${JSON.stringify(srflxCandidates)}`);
+  // TODO(bug 1339203): Once we support rtcpMuxPolicy, set it to "require" to
+  // result in a single srflx candidate
+  is(
+    srflxCandidates.length,
+    2,
+    `Should have two srflx candidates with ${context}`
+  );
+  pc.close();
+}
+
+async function checkNoSrflx(iceServers) {
+  const context = makeContextString(iceServers);
+  info(`checkNoSrflx ${context}`);
+  const pc = new RTCPeerConnection({
+    iceServers,
+    bundlePolicy: "max-bundle", // Avoids extra candidates
+  });
+  const candidates = await gatherWithTimeout(pc, 8000, context);
+  const srflxCandidates = candidates.filter(c => c.candidate.includes("srflx"));
+  info(`candidates: ${JSON.stringify(srflxCandidates)}`);
+  is(
+    srflxCandidates.length,
+    0,
+    `Should have no srflx candidates with ${context}`
+  );
+  pc.close();
+}
+
+async function checkRelayUdp(iceServers) {
+  const context = makeContextString(iceServers);
+  info(`checkRelayUdp ${context}`);
+  const pc = new RTCPeerConnection({
+    iceServers,
+    bundlePolicy: "max-bundle", // Avoids extra candidates
+  });
+  const candidates = await gatherWithTimeout(pc, 8000, context);
+  const relayCandidates = candidates.filter(c => c.candidate.includes("relay"));
+  info(`candidates: ${JSON.stringify(relayCandidates)}`);
+  // TODO(bug 1339203): Once we support rtcpMuxPolicy, set it to "require" to
+  // result in a single relay candidate
+  is(
+    relayCandidates.length,
+    2,
+    `Should have two relay candidates with ${context}`
+  );
+  // It would be nice if RTCIceCandidate had a field telling us what the
+  // "related protocol" is (similar to relatedAddress and relatedPort).
+  // Because there is no such thing, we need to go through the stats API,
+  // which _does_ have that information.
+  is(
+    (await findStatsRelayCandidates(pc, "tcp")).length,
+    0,
+    `No TCP relay candidates should be present with ${context}`
+  );
+  pc.close();
+}
+
+async function checkRelayTcp(iceServers) {
+  const context = makeContextString(iceServers);
+  info(`checkRelayTcp ${context}`);
+  const pc = new RTCPeerConnection({
+    iceServers,
+    bundlePolicy: "max-bundle", // Avoids extra candidates
+  });
+  const candidates = await gatherWithTimeout(pc, 8000, context);
+  const relayCandidates = candidates.filter(c => c.candidate.includes("relay"));
+  info(`candidates: ${JSON.stringify(relayCandidates)}`);
+  // TODO(bug 1339203): Once we support rtcpMuxPolicy, set it to "require" to
+  // result in a single relay candidate
+  is(
+    relayCandidates.length,
+    2,
+    `Should have two relay candidates with ${context}`
+  );
+  // It would be nice if RTCIceCandidate had a field telling us what the
+  // "related protocol" is (similar to relatedAddress and relatedPort).
+  // Because there is no such thing, we need to go through the stats API,
+  // which _does_ have that information.
+  is(
+    (await findStatsRelayCandidates(pc, "udp")).length,
+    0,
+    `No UDP relay candidates should be present with ${context}`
+  );
+  pc.close();
+}
+
+async function checkRelayUdpTcp(iceServers) {
+  const context = makeContextString(iceServers);
+  info(`checkRelayUdpTcp ${context}`);
+  const pc = new RTCPeerConnection({
+    iceServers,
+    bundlePolicy: "max-bundle", // Avoids extra candidates
+  });
+  const candidates = await gatherWithTimeout(pc, 8000, context);
+  const relayCandidates = candidates.filter(c => c.candidate.includes("relay"));
+  info(`candidates: ${JSON.stringify(relayCandidates)}`);
+  // TODO(bug 1339203): Once we support rtcpMuxPolicy, set it to "require" to
+  // result in a single relay candidate each for UDP and TCP
+  is(
+    relayCandidates.length,
+    4,
+    `Should have two relay candidates for each protocol with ${context}`
+  );
+  // It would be nice if RTCIceCandidate had a field telling us what the
+  // "related protocol" is (similar to relatedAddress and relatedPort).
+  // Because there is no such thing, we need to go through the stats API,
+  // which _does_ have that information.
+  is(
+    (await findStatsRelayCandidates(pc, "udp")).length,
+    2,
+    `Two UDP relay candidates should be present with ${context}`
+  );
+  // TODO(bug 1705563): This is 1 because of bug 1705563
+  is(
+    (await findStatsRelayCandidates(pc, "tcp")).length,
+    1,
+    `One TCP relay candidates should be present with ${context}`
+  );
+  pc.close();
+}
+
+async function checkNoRelay(iceServers) {
+  const context = makeContextString(iceServers);
+  info(`checkNoRelay ${context}`);
+  const pc = new RTCPeerConnection({
+    iceServers,
+    bundlePolicy: "max-bundle", // Avoids extra candidates
+  });
+  const candidates = await gatherWithTimeout(pc, 8000, context);
+  const relayCandidates = candidates.filter(c => c.candidate.includes("relay"));
+  info(`candidates: ${JSON.stringify(relayCandidates)}`);
+  is(
+    relayCandidates.length,
+    0,
+    `Should have no relay candidates with ${context}`
+  );
+  pc.close();
+}
diff --git a/dom/media/webrtc/tests/mochitests/identity/identityPcTest.js b/dom/media/webrtc/tests/mochitests/identity/identityPcTest.js
new file mode 100644
index 0000000000..1381873f9d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/identityPcTest.js
@@ -0,0 +1,79 @@
+function identityPcTest(remoteOptions) {
+  var user = "someone";
+  var domain1 = "test1.example.com";
+  var domain2 = "test2.example.com";
+  var id1 = user + "@" + domain1;
+  var id2 = user + "@" + domain2;
+
+  test = new PeerConnectionTest({
+    config_local: {
+      peerIdentity: id2,
+    },
+    config_remote: {
+      peerIdentity: id1,
+    },
+  });
+  test.setMediaConstraints(
+    [
+      {
+        audio: true,
+        video: true,
+        peerIdentity: id2,
+      },
+    ],
+    [
+      remoteOptions || {
+        audio: true,
+        video: true,
+        peerIdentity: id1,
+      },
+    ]
+  );
+  test.pcLocal.setIdentityProvider("test1.example.com", { protocol: "idp.js" });
+  test.pcRemote.setIdentityProvider("test2.example.com", {
+    protocol: "idp.js",
+  });
+  test.chain.append([
+    function PEER_IDENTITY_IS_SET_CORRECTLY(test) {
+      // no need to wait to check identity in this case,
+      // setRemoteDescription should wait for the IdP to complete
+      function checkIdentity(pc, pfx, idp, name) {
+        return pc.peerIdentity.then(peerInfo => {
+          is(peerInfo.idp, idp, pfx + "IdP check");
+          is(peerInfo.name, name + "@" + idp, pfx + "identity check");
+        });
+      }
+
+      return Promise.all([
+        checkIdentity(
+          test.pcLocal._pc,
+          "local: ",
+          "test2.example.com",
+          "someone"
+        ),
+        checkIdentity(
+          test.pcRemote._pc,
+          "remote: ",
+          "test1.example.com",
+          "someone"
+        ),
+      ]);
+    },
+
+    function REMOTE_STREAMS_ARE_RESTRICTED(test) {
+      var remoteStream = test.pcLocal._pc.getRemoteStreams()[0];
+      for (const track of remoteStream.getTracks()) {
+        mustThrowWith(
+          `Freshly received ${track.kind} track with peerIdentity`,
+          "SecurityError",
+          () => new MediaRecorder(new MediaStream([track])).start()
+        );
+      }
+      return Promise.all([
+        audioIsSilence(true, remoteStream),
+        videoIsBlack(true, remoteStream),
+      ]);
+    },
+  ]);
+  return test.run();
+}
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-bad.js b/dom/media/webrtc/tests/mochitests/identity/idp-bad.js
new file mode 100644
index 0000000000..86e1cb7a34
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-bad.js
@@ -0,0 +1 @@
+<This isn't valid JS>
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-min.js b/dom/media/webrtc/tests/mochitests/identity/idp-min.js
new file mode 100644
index 0000000000..a4b2c55cee
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-min.js
@@ -0,0 +1,24 @@
+(function (global) {
+  "use strict";
+  // A minimal implementation of the interface.
+  // Though this isn't particularly functional.
+  // This is needed so that we can have a "working" IdP served
+  // from two different locations in the tree.
+  global.rtcIdentityProvider.register({
+    generateAssertion(payload, origin, usernameHint) {
+      dump("idp: generateAssertion(" + payload + ")\n");
+      return Promise.resolve({
+        idp: { domain: "example.com", protocol: "idp.js" },
+        assertion: "bogus",
+      });
+    },
+
+    validateAssertion(assertion, origin) {
+      dump("idp: validateAssertion(" + assertion + ")\n");
+      return Promise.resolve({
+        identity: "user@example.com",
+        contents: "bogus",
+      });
+    },
+  });
+})(this);
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http-trick.js b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http-trick.js
new file mode 100644
index 0000000000..75390cbf4f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http-trick.js
@@ -0,0 +1,3 @@
+(function () {
+  dump("ERROR\n");
+})();
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http-trick.js^headers^ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http-trick.js^headers^
new file mode 100644
index 0000000000..b3a2afd90a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http-trick.js^headers^
@@ -0,0 +1,2 @@
+HTTP 301 Moved Permanently
+Location: http://example.com/.well-known/idp-proxy/idp-redirect-https.js
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http.js b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http.js
new file mode 100644
index 0000000000..75390cbf4f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http.js
@@ -0,0 +1,3 @@
+(function () {
+  dump("ERROR\n");
+})();
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http.js^headers^ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http.js^headers^
new file mode 100644
index 0000000000..d2380984e7
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http.js^headers^
@@ -0,0 +1,2 @@
+HTTP 301 Moved Permanently
+Location: http://example.com/.well-known/idp-proxy/idp.js
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-double.js b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-double.js
new file mode 100644
index 0000000000..75390cbf4f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-double.js
@@ -0,0 +1,3 @@
+(function () {
+  dump("ERROR\n");
+})();
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-double.js^headers^ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-double.js^headers^
new file mode 100644
index 0000000000..3fb8a35ae7
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-double.js^headers^
@@ -0,0 +1,2 @@
+HTTP 301 Moved Permanently
+Location: https://example.com/.well-known/idp-proxy/idp-redirect-https.js
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-odd-path.js b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-odd-path.js
new file mode 100644
index 0000000000..75390cbf4f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-odd-path.js
@@ -0,0 +1,3 @@
+(function () {
+  dump("ERROR\n");
+})();
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-odd-path.js^headers^ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-odd-path.js^headers^
new file mode 100644
index 0000000000..6e2931eda9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-odd-path.js^headers^
@@ -0,0 +1,2 @@
+HTTP 301 Moved Permanently
+Location: https://example.com/.well-known/idp-min.js
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https.js b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https.js
new file mode 100644
index 0000000000..75390cbf4f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https.js
@@ -0,0 +1,3 @@
+(function () {
+  dump("ERROR\n");
+})();
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https.js^headers^ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https.js^headers^
new file mode 100644
index 0000000000..77d56ac442
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https.js^headers^
@@ -0,0 +1,2 @@
+HTTP 301 Moved Permanently
+Location: https://example.com/.well-known/idp-proxy/idp.js
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp.js b/dom/media/webrtc/tests/mochitests/identity/idp.js
new file mode 100644
index 0000000000..557740657f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp.js
@@ -0,0 +1,119 @@
+(function (global) {
+  "use strict";
+
+  // rather than create a million different IdP configurations and litter the
+  // world with files all containing near-identical code, let's use the hash/URL
+  // fragment as a way of generating instructions for the IdP
+  var instructions = global.location.hash.replace("#", "").split(":");
+  function is(target) {
+    return function (instruction) {
+      return instruction === target;
+    };
+  }
+
+  function IDPJS() {
+    this.domain = global.location.host;
+    var path = global.location.pathname;
+    this.protocol =
+      path.substring(path.lastIndexOf("/") + 1) + global.location.hash;
+    this.id = crypto.getRandomValues(new Uint8Array(10)).join(".");
+  }
+
+  IDPJS.prototype = {
+    getLogin() {
+      return fetch(
+        "https://example.com/.well-known/idp-proxy/idp.sjs?" + this.id
+      ).then(response => response.status === 200);
+    },
+    checkLogin(result) {
+      return this.getLogin().then(loggedIn => {
+        if (loggedIn) {
+          return result;
+        }
+        return Promise.reject({
+          name: "IdpLoginError",
+          loginUrl:
+            "https://example.com/.well-known/idp-proxy/login.html#" + this.id,
+        });
+      });
+    },
+
+    borkResult(result) {
+      if (instructions.some(is("throw"))) {
+        throw new Error("Throwing!");
+      }
+      if (instructions.some(is("fail"))) {
+        return Promise.reject(new Error("Failing!"));
+      }
+      if (instructions.some(is("login"))) {
+        return this.checkLogin(result);
+      }
+      if (instructions.some(is("hang"))) {
+        return new Promise(r => {});
+      }
+      dump("idp: result=" + JSON.stringify(result) + "\n");
+      return Promise.resolve(result);
+    },
+
+    _selectUsername(usernameHint) {
+      dump("_selectUsername: usernameHint(" + usernameHint + ")\n");
+      var username = "someone@" + this.domain;
+      if (usernameHint) {
+        var at = usernameHint.indexOf("@");
+        if (at < 0) {
+          username = usernameHint + "@" + this.domain;
+        } else if (usernameHint.substring(at + 1) === this.domain) {
+          username = usernameHint;
+        }
+      }
+      return username;
+    },
+
+    generateAssertion(payload, origin, options) {
+      dump(
+        "idp: generateAssertion(" +
+          payload +
+          ", " +
+          origin +
+          ", " +
+          JSON.stringify(options) +
+          ")\n"
+      );
+      var idpDetails = {
+        domain: this.domain,
+        protocol: this.protocol,
+      };
+      if (instructions.some(is("bad-assert"))) {
+        idpDetails = {};
+      }
+      return this.borkResult({
+        idp: idpDetails,
+        assertion: JSON.stringify({
+          username: this._selectUsername(options.usernameHint),
+          contents: payload,
+        }),
+      });
+    },
+
+    validateAssertion(assertion, origin) {
+      dump("idp: validateAssertion(" + assertion + ")\n");
+      var assertion = JSON.parse(assertion);
+      if (instructions.some(is("bad-validate"))) {
+        assertion.contents = {};
+      }
+      return this.borkResult({
+        identity: assertion.username,
+        contents: assertion.contents,
+      });
+    },
+  };
+
+  if (!instructions.some(is("not_ready"))) {
+    dump("registering idp.js" + global.location.hash + "\n");
+    var idp = new IDPJS();
+    global.rtcIdentityProvider.register({
+      generateAssertion: idp.generateAssertion.bind(idp),
+      validateAssertion: idp.validateAssertion.bind(idp),
+    });
+  }
+})(this);
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp.sjs b/dom/media/webrtc/tests/mochitests/identity/idp.sjs
new file mode 100644
index 0000000000..e1a245be78
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp.sjs
@@ -0,0 +1,18 @@
+function handleRequest(request, response) {
+  var key = "/.well-known/idp-proxy/" + request.queryString;
+  dump(getState(key) + "\n");
+  if (request.method === "GET") {
+    if (getState(key)) {
+      response.setStatusLine(request.httpVersion, 200, "OK");
+    } else {
+      response.setStatusLine(request.httpVersion, 404, "Not Found");
+    }
+  } else if (request.method === "PUT") {
+    setState(key, "OK");
+    response.setStatusLine(request.httpVersion, 200, "OK");
+  } else {
+    response.setStatusLine(request.httpVersion, 406, "Method Not Allowed");
+  }
+  response.setHeader("Content-Type", "text/plain;charset=UTF-8");
+  response.write("OK");
+}
diff --git a/dom/media/webrtc/tests/mochitests/identity/login.html b/dom/media/webrtc/tests/mochitests/identity/login.html
new file mode 100644
index 0000000000..eafba22f2d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/login.html
@@ -0,0 +1,31 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Identity Provider Login</title>
+  <script type="application/javascript">
+  window.onload = () => {
+    var xhr = new XMLHttpRequest();
+    xhr.open("PUT", "https://example.com/.well-known/idp-proxy/idp.sjs?" +
+             window.location.hash.replace('#', ''));
+    xhr.onload = () => {
+      var isFramed = (window !== window.top);
+      var parent = isFramed ? window.parent : window.opener;
+      // Using '*' is cheating, but that's OK.
+      parent.postMessage('LOGINDONE', '*');
+      var done = document.createElement('div');
+
+      done.textContent = 'Done';
+      document.body.appendChild(done);
+
+      if (!isFramed) {
+        window.close();
+      }
+    };
+    xhr.send();
+  };
+  </script>
+</head>
+<body>
+  <div>Logging in...</div>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/mochitest.ini b/dom/media/webrtc/tests/mochitests/identity/mochitest.ini
new file mode 100644
index 0000000000..7a60cc6c6e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/mochitest.ini
@@ -0,0 +1,47 @@
+[DEFAULT]
+subsuite = media
+skip-if = (os == 'linux' && !debug)
+support-files =
+  /.well-known/idp-proxy/idp.js
+  identityPcTest.js
+  !/dom/media/webrtc/tests/mochitests/blacksilence.js
+  !/dom/media/webrtc/tests/mochitests/dataChannel.js
+  !/dom/media/webrtc/tests/mochitests/head.js
+  !/dom/media/webrtc/tests/mochitests/network.js
+  !/dom/media/webrtc/tests/mochitests/pc.js
+  !/dom/media/webrtc/tests/mochitests/sdpUtils.js
+  !/dom/media/webrtc/tests/mochitests/templates.js
+  !/dom/media/webrtc/tests/mochitests/turnConfig.js
+tags = mtg
+
+[test_fingerprints.html]
+scheme=https
+[test_getIdentityAssertion.html]
+[test_idpproxy.html]
+support-files =
+  /.well-known/idp-proxy/idp-redirect-http.js
+  /.well-known/idp-proxy/idp-redirect-http.js^headers^
+  /.well-known/idp-proxy/idp-redirect-http-trick.js
+  /.well-known/idp-proxy/idp-redirect-http-trick.js^headers^
+  /.well-known/idp-proxy/idp-redirect-https.js
+  /.well-known/idp-proxy/idp-redirect-https.js^headers^
+  /.well-known/idp-proxy/idp-redirect-https-double.js
+  /.well-known/idp-proxy/idp-redirect-https-double.js^headers^
+  /.well-known/idp-proxy/idp-redirect-https-odd-path.js
+  /.well-known/idp-proxy/idp-redirect-https-odd-path.js^headers^
+  /.well-known/idp-min.js
+  /.well-known/idp-proxy/idp-bad.js
+[test_loginNeeded.html]
+support-files =
+  /.well-known/idp-proxy/login.html
+  /.well-known/idp-proxy/idp.sjs
+[test_peerConnection_asymmetricIsolation.html]
+scheme=https
+skip-if = os == 'android'
+[test_peerConnection_peerIdentity.html]
+scheme=https
+skip-if = os == 'android'
+[test_setIdentityProvider.html]
+scheme=https
+[test_setIdentityProviderWithErrors.html]
+scheme=https
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_fingerprints.html b/dom/media/webrtc/tests/mochitests/identity/test_fingerprints.html
new file mode 100644
index 0000000000..0a7f0a2033
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_fingerprints.html
@@ -0,0 +1,91 @@
+<html>
+<head>
+<meta charset="utf-8" />
+<script type="application/javascript">var scriptRelativePath = "../";</script>
+<script type="application/javascript" src="../pc.js"></script>
+</head>
+<body>
+<script class="testbody" type="application/javascript">
+createHTML({ title: "Test multiple identity fingerprints",  bug: "1005152" });
+
+// here we call the identity provider directly
+async function getIdentityAssertion(fingerprint) {
+  const { IdpSandbox } = SpecialPowers.ChromeUtils.import(
+    'resource://gre/modules/media/IdpSandbox.jsm'
+  );
+  const sandbox = new IdpSandbox('example.com', 'idp.js', window);
+  const idp = SpecialPowers.wrap(await sandbox.start());
+  const assertion = SpecialPowers.wrap(await
+      idp.generateAssertion(JSON.stringify({ fingerprint }),
+                            'https://example.com',
+                            {}));
+  const assertionString = btoa(JSON.stringify(assertion));
+  sandbox.stop();
+  return assertionString;
+}
+
+// This takes a real fingerprint and makes some extra bad ones.
+function makeFingerprints(algorithm, digest) {
+  const fingerprints = [];
+  fingerprints.push({ algorithm, digest });
+  for (var i = 0; i < 3; ++i) {
+    fingerprints.push({
+      algorithm,
+      digest: digest.replace(/:./g, ':' + i.toString(16))
+    });
+  }
+  return fingerprints;
+}
+
+const fingerprintRegex = /^a=fingerprint:(\S+) (\S+)/m;
+const identityRegex = /^a=identity:(\S+)/m;
+
+function fingerprintSdp(fingerprints) {
+  return fingerprints.map(fp => 'a=fInGeRpRiNt:' + fp.algorithm +
+                                ' ' + fp.digest + '\n').join('');
+}
+
+// Firefox only uses a single fingerprint.
+// That doesn't mean we have it create SDP that describes two.
+// This function synthesizes that SDP and tries to set it.
+
+runNetworkTest(async () => {
+  // this one fails setRemoteDescription if the identity is not good
+  const pcStrict = new RTCPeerConnection({ peerIdentity: 'someone@example.com'});
+  // this one will be manually tweaked to have two fingerprints
+  const pcDouble = new RTCPeerConnection({});
+
+  const stream = await getUserMedia({ video: true });
+  ok(stream, 'Got test stream');
+  const [track] = stream.getTracks();
+  pcDouble.addTrack(track, stream);
+  try {
+    const offer = await pcDouble.createOffer();
+    ok(offer, 'Got offer');
+    const match = offer.sdp.match(fingerprintRegex);
+    if (!match) {
+      throw new Error('No fingerprint in offer SDP');
+    }
+    const fingerprints = makeFingerprints(match[1], match[2]);
+    const assertion = await getIdentityAssertion(fingerprints);
+    ok(assertion, 'Should have assertion');
+
+    const sdp = offer.sdp.slice(0, match.index) +
+        'a=identity:' + assertion + '\n' +
+        fingerprintSdp(fingerprints.slice(1)) +
+        offer.sdp.slice(match.index);
+
+    await pcStrict.setRemoteDescription({ type: 'offer', sdp });
+    ok(true, 'Modified fingerprints were accepted');
+  } catch (error) {
+    const e = SpecialPowers.wrap(error);
+    ok(false, 'error in test: ' +
+       (e.message ? (e.message + '\n' + e.stack) : e));
+  }
+  pcStrict.close();
+  pcDouble.close();
+  track.stop();
+});
+</script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_getIdentityAssertion.html b/dom/media/webrtc/tests/mochitests/identity/test_getIdentityAssertion.html
new file mode 100644
index 0000000000..47e1cb1df6
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_getIdentityAssertion.html
@@ -0,0 +1,101 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript">var scriptRelativePath = "../";</script>
+  <script type="application/javascript" src="../pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getIdentityAssertion Tests",
+    bug: "942367"
+  });
+
+function checkIdentity(assertion, identity) {
+  // here we dig into the payload, which means we need to know something
+  // about how the IdP actually works (not good in general, but OK here)
+  var assertion = JSON.parse(atob(assertion)).assertion;
+  var user = JSON.parse(assertion).username;
+  is(user, identity, 'id should be "' + identity + '" is "' + user + '"');
+}
+
+function getAssertion(t, instructions, userHint) {
+  dump('instructions: ' + instructions + '\n');
+  dump('userHint: ' + userHint + '\n');
+  t.pcLocal.setIdentityProvider('example.com',
+                                { protocol: 'idp.js' + instructions,
+                                  usernameHint: userHint });
+  return t.pcLocal._pc.getIdentityAssertion();
+}
+
+var test;
+function theTest() {
+  test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter('PC_REMOTE_CHECK_INITIAL_SIGNALINGSTATE');
+  test.chain.append([
+    function PC_LOCAL_IDENTITY_ASSERTION_FAILS_WITHOUT_PROVIDER(t) {
+      return t.pcLocal._pc.getIdentityAssertion()
+        .then(a => ok(false, 'should fail without provider'),
+              e => ok(e, 'should fail without provider'));
+    },
+
+    function PC_LOCAL_IDENTITY_ASSERTION_FAILS_WITH_BAD_PROVIDER(t) {
+      t.pcLocal._pc.setIdentityProvider('example.com',
+                                        { protocol: 'idp-bad.js',
+                                          usernameHint: '' });
+      return t.pcLocal._pc.getIdentityAssertion()
+        .then(a => ok(false, 'should fail with bad provider'),
+              e => {
+                is(e.name, 'IdpError', 'should fail with bad provider');
+                ok(e.message, 'should include a nice message');
+              });
+    },
+
+    function PC_LOCAL_GET_TWO_ASSERTIONS(t) {
+      return Promise.all([
+        getAssertion(t, ''),
+        getAssertion(t, '')
+      ]).then(assertions => {
+        is(assertions.length, 2, "Two assertions generated");
+        assertions.forEach(a => checkIdentity(a, 'someone@example.com'));
+      });
+    },
+
+    function PC_LOCAL_IDP_FAILS(t) {
+      return getAssertion(t, '#fail')
+        .then(a => ok(false, '#fail should not get an identity result'),
+              e => is(e.name, 'IdpError', '#fail should cause rejection'));
+    },
+
+    function PC_LOCAL_IDP_LOGIN_ERROR(t) {
+      return getAssertion(t, '#login')
+        .then(a => ok(false, '#login should not work'),
+              e => {
+                is(e.name, 'IdpLoginError', 'name is IdpLoginError');
+                is(t.pcLocal._pc.idpLoginUrl.split('#')[0],
+                   'https://example.com/.well-known/idp-proxy/login.html',
+                   'got the right login URL from the IdP');
+              });
+    },
+
+    function PC_LOCAL_IDP_NOT_READY(t) {
+      return getAssertion(t, '#not_ready')
+        .then(a => ok(false, '#not_ready should not get an identity result'),
+              e => is(e.name, 'IdpError', '#not_ready should cause rejection'));
+    },
+
+    function PC_LOCAL_ASSERTION_WITH_SPECIFIC_NAME(t) {
+      return getAssertion(t, '', 'user@example.com')
+        .then(a => checkIdentity(a, 'user@example.com'));
+    }
+  ]);
+  return test.run();
+}
+runNetworkTest(theTest);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_idpproxy.html b/dom/media/webrtc/tests/mochitests/identity/test_idpproxy.html
new file mode 100644
index 0000000000..065501b8a4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_idpproxy.html
@@ -0,0 +1,178 @@
+<html>
+<head>
+<meta charset="utf-8" />
+<link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+<script src="/tests/SimpleTest/SimpleTest.js"></script>
+</head>
+<body>
+  <script class="testbody" type="application/javascript">
+"use strict";
+var { IdpSandbox } = SpecialPowers.ChromeUtils.import(
+  "resource://gre/modules/media/IdpSandbox.jsm"
+);
+var dummyPayload = JSON.stringify({
+  this: 'is',
+  a: ['stu', 6],
+  obj: null
+});
+
+function test_domain_sandbox() {
+  var diabolical = {
+    toString() {
+      return 'example.com/path';
+    }
+  };
+  var domains = [ 'ex/foo', 'user@ex', 'user:pass@ex', 'ex#foo', 'ex?foo',
+                  '', 12, null, diabolical, true ];
+  domains.forEach(function(domain) {
+    try {
+      var idp = new IdpSandbox(domain, undefined, window);
+      ok(false, 'IdpSandbox allowed a bad domain: ' + domain);
+    } catch (e) {
+      var str = (typeof domain === 'string') ? domain : typeof domain;
+      ok(true, 'Evil domain "' + str + '" raises exception');
+    }
+  });
+}
+
+function test_protocol_sandbox() {
+  var protos = [ '../evil/proto', '..%2Fevil%2Fproto',
+                 '\\evil', '%5cevil', 12, true, {} ];
+  protos.forEach(function(proto) {
+    try {
+      var idp = new IdpSandbox('example.com', proto, window);
+      ok(false, 'IdpSandbox allowed a bad protocol: ' + proto);
+    } catch (e) {
+      var str = (typeof proto === 'string') ? proto : typeof proto;
+      ok(true, 'Evil protocol "' + proto + '" raises exception');
+    }
+  });
+}
+
+function idpName(hash) {
+  return 'idp.js' + (hash ? ('#' + hash) : '');
+}
+
+function makeSandbox(js) {
+  var name = js || idpName();
+  info('Creating a sandbox for the protocol: ' + name);
+  var sandbox = new IdpSandbox('example.com', name, window);
+  return sandbox.start().then(idp => SpecialPowers.wrap(idp));
+}
+
+function test_generate_assertion() {
+  return makeSandbox()
+    .then(idp => idp.generateAssertion(dummyPayload,
+                                       'https://example.net',
+                                        {}))
+    .then(response => {
+      response = SpecialPowers.wrap(response);
+      is(response.idp.domain, 'example.com', 'domain is correct');
+      is(response.idp.protocol, 'idp.js', 'protocol is correct');
+      ok(typeof response.assertion === 'string', 'assertion is present');
+    });
+}
+
+// test that the test IdP can eat its own dogfood; which is the only way to test
+// validateAssertion, since that consumes the output of generateAssertion (in
+// theory, generateAssertion could identify a different IdP domain).
+
+function test_validate_assertion() {
+  return makeSandbox()
+    .then(idp => idp.generateAssertion(dummyPayload,
+                                       'https://example.net',
+                                       { usernameHint: 'user' }))
+    .then(assertion => {
+      var wrapped = SpecialPowers.wrap(assertion);
+      return makeSandbox()
+        .then(idp => idp.validateAssertion(wrapped.assertion,
+                                           'https://example.net'));
+    }).then(response => {
+      response = SpecialPowers.wrap(response);
+      is(response.identity, 'user@example.com');
+      is(response.contents, dummyPayload);
+    });
+}
+
+// We don't want to test the #bad or the #hang instructions,
+// errors of the sort those generate aren't handled by the sandbox code.
+function test_assertion_failure(reason) {
+  return () => {
+    return makeSandbox(idpName(reason))
+      .then(idp => idp.generateAssertion('hello', 'example.net', {}))
+      .then(r => ok(false, 'should not succeed on ' + reason),
+            e => ok(true, 'failed correctly on ' + reason));
+  };
+}
+
+function test_load_failure() {
+  return makeSandbox('non-existent-file')
+    .then(() => ok(false, 'Should fail to load non-existent file'),
+          e => ok(e, 'Should fail to load non-existent file'));
+}
+
+function test_redirect_ok(from) {
+  return () => {
+    return makeSandbox(from)
+      .then(idp => idp.generateAssertion('hello', 'example.net'))
+      .then(r => ok(SpecialPowers.wrap(r).assertion,
+                    'Redirect to https should be OK'));
+  };
+}
+
+function test_redirect_fail(from) {
+  return () => {
+    return makeSandbox(from)
+      .then(() => ok(false, 'Redirect to https should fail'),
+            e => ok(e, 'Redirect to https should fail'));
+  };
+}
+
+function test_bad_js() {
+  return makeSandbox('idp-bad.js')
+    .then(() => ok(false, 'Bad JS should not load'),
+          e => ok(e, 'Bad JS should not load'));
+}
+
+function run_all_tests() {
+  [
+    test_domain_sandbox,
+    test_protocol_sandbox,
+    test_generate_assertion,
+    test_validate_assertion,
+
+    // fail of the IdP fails
+    test_assertion_failure('fail'),
+    // fail if the IdP throws
+    test_assertion_failure('throw'),
+    // fail if the IdP is not ready
+    test_assertion_failure('not_ready'),
+
+    test_load_failure(),
+    // Test a redirect to an HTTPS origin, which should be OK
+    test_redirect_ok('idp-redirect-https.js'),
+    // Two redirects is fine too
+    test_redirect_ok('idp-redirect-https-double.js'),
+    // A secure redirect to a path other than /.well-known/idp-proxy/* should
+    // also work fine.
+    test_redirect_ok('idp-redirect-https-odd-path.js'),
+    // A redirect to HTTP is not-cool
+    test_redirect_fail('idp-redirect-http.js'),
+    // Also catch tricks like https->http->https
+    test_redirect_fail('idp-redirect-http-trick.js'),
+
+    test_bad_js
+  ].reduce((p, test) => {
+    return p.then(test)
+      .catch(e => ok(false, test.name + ' failed: ' +
+                     SpecialPowers.wrap(e).message + '\n' +
+                     SpecialPowers.wrap(e).stack));
+  }, Promise.resolve())
+    .then(() => SimpleTest.finish());
+}
+
+SimpleTest.waitForExplicitFinish();
+run_all_tests();
+</script>
+  </body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_loginNeeded.html b/dom/media/webrtc/tests/mochitests/identity/test_loginNeeded.html
new file mode 100644
index 0000000000..550dc20d92
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_loginNeeded.html
@@ -0,0 +1,72 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript">var scriptRelativePath = "../";</script>
+  <script type="application/javascript" src="../pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: 'RTCPeerConnection identity with login',
+    bug: '1153314'
+  });
+
+function waitForLoginDone() {
+  return new Promise(resolve => {
+    window.addEventListener('message', function listener(e) {
+      is(e.origin, 'https://example.com', 'got the right message origin');
+      is(e.data, 'LOGINDONE', 'got the right message');
+      window.removeEventListener('message', listener);
+      resolve();
+    });
+  });
+}
+
+function checkLogin(t, name, onLoginNeeded) {
+  t.pcLocal.setIdentityProvider('example.com',
+                                { protocol: 'idp.js#login:' + name });
+  return t.pcLocal._pc.getIdentityAssertion()
+    .then(a => ok(false, 'should request login'),
+          e => {
+            is(e.name, 'IdpLoginError', 'name is IdpLoginError');
+            is(t.pcLocal._pc.idpLoginUrl.split('#')[0],
+               'https://example.com/.well-known/idp-proxy/login.html',
+               'got the right login URL from the IdP');
+            return t.pcLocal._pc.idpLoginUrl;
+          })
+    .then(onLoginNeeded)
+    .then(waitForLoginDone)
+    .then(() => t.pcLocal._pc.getIdentityAssertion())
+    .then(a => ok(a, 'got assertion'));
+}
+
+function theTest() {
+  var test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter('PC_REMOTE_CHECK_INITIAL_SIGNALINGSTATE');
+  test.chain.append([
+    function PC_LOCAL_IDENTITY_ASSERTION_WITH_IFRAME_LOGIN(t) {
+      return checkLogin(t, 'iframe', loginUrl => {
+        var iframe = document.createElement('iframe');
+        iframe.setAttribute('src', loginUrl);
+        iframe.frameBorder = 0;
+        iframe.width = 400;
+        iframe.height = 60;
+        document.getElementById('display').appendChild(iframe);
+      });
+    },
+    function PC_LOCAL_IDENTITY_ASSERTION_WITH_WINDOW_LOGIN(t) {
+      return checkLogin(t, 'openwin', loginUrl => {
+        window.open(loginUrl, 'login', 'width=400,height=60');
+      });
+    }
+  ]);
+  return test.run();
+}
+runNetworkTest(theTest);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_peerConnection_asymmetricIsolation.html b/dom/media/webrtc/tests/mochitests/identity/test_peerConnection_asymmetricIsolation.html
new file mode 100644
index 0000000000..65a2fc5392
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_peerConnection_asymmetricIsolation.html
@@ -0,0 +1,31 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript">var scriptRelativePath = "../";</script>
+  <script type="application/javascript" src="../pc.js"></script>
+  <script type="application/javascript" src="../blacksilence.js"></script>
+  <script type="application/javascript" src="identityPcTest.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "Non-isolated media entering an isolated session becomes isolated",
+  bug: "996238"
+});
+
+function theTest() {
+  // Override the remote media capture options to remove isolation for the
+  // remote party; the test verifies that the media it receives on the local
+  // side is isolated anyway.
+  return identityPcTest({
+    audio: true,
+    video: true
+  });
+}
+runNetworkTest(theTest);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_peerConnection_peerIdentity.html b/dom/media/webrtc/tests/mochitests/identity/test_peerConnection_peerIdentity.html
new file mode 100644
index 0000000000..a8116cc451
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_peerConnection_peerIdentity.html
@@ -0,0 +1,21 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript">var scriptRelativePath = "../";</script>
+  <script type="application/javascript" src="../pc.js"></script>
+  <script type="application/javascript" src="../blacksilence.js"></script>
+  <script type="application/javascript" src="identityPcTest.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "setIdentityProvider leads to peerIdentity and assertions in SDP",
+  bug: "942367"
+});
+
+runNetworkTest(identityPcTest);
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_setIdentityProvider.html b/dom/media/webrtc/tests/mochitests/identity/test_setIdentityProvider.html
new file mode 100644
index 0000000000..ac7cba6a5e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_setIdentityProvider.html
@@ -0,0 +1,67 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript">var scriptRelativePath = "../";</script>
+  <script type="application/javascript" src="../pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "setIdentityProvider leads to peerIdentity and assertions in SDP",
+    bug: "942367"
+  });
+
+function checkIdentity(peer, prefix, idp, name) {
+  prefix = prefix + ": ";
+  return peer._pc.peerIdentity.then(peerIdentity => {
+    ok(peerIdentity, prefix + "peerIdentity is set");
+    is(peerIdentity.idp, idp, prefix + "IdP is correct");
+    is(peerIdentity.name, name + "@" + idp, prefix + "identity is correct");
+  });
+}
+
+function theTest() {
+  var test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.pcLocal.setIdentityProvider("test1.example.com",
+                                   { protocol: "idp.js",
+                                     usernameHint: "someone" });
+  test.pcRemote.setIdentityProvider("test2.example.com",
+                                    { protocol: "idp.js",
+                                      usernameHinte: "someone"});
+
+  test.chain.append([
+    function PC_LOCAL_PEER_IDENTITY_IS_SET_CORRECTLY(test) {
+      return checkIdentity(test.pcLocal, "local", "test2.example.com", "someone");
+    },
+    function PC_REMOTE_PEER_IDENTITY_IS_SET_CORRECTLY(test) {
+      return checkIdentity(test.pcRemote, "remote", "test1.example.com", "someone");
+    },
+
+    function OFFER_AND_ANSWER_INCLUDES_IDENTITY(test) {
+      ok(test.originalOffer.sdp.includes("a=identity"), "a=identity is in the offer SDP");
+      ok(test.originalAnswer.sdp.includes("a=identity"), "a=identity is in the answer SDP");
+    },
+
+    function PC_LOCAL_DESCRIPTIONS_CONTAIN_IDENTITY(test) {
+      ok(test.pcLocal.localDescription.sdp.includes("a=identity"),
+         "a=identity is in the local copy of the offer");
+      ok(test.pcLocal.remoteDescription.sdp.includes("a=identity"),
+         "a=identity is in the local copy of the answer");
+    },
+    function PC_REMOTE_DESCRIPTIONS_CONTAIN_IDENTITY(test) {
+      ok(test.pcRemote.localDescription.sdp.includes("a=identity"),
+                         "a=identity is in the remote copy of the offer");
+      ok(test.pcRemote.remoteDescription.sdp.includes("a=identity"),
+                         "a=identity is in the remote copy of the answer");
+    }
+  ]);
+  return test.run();
+}
+runNetworkTest(theTest);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_setIdentityProviderWithErrors.html b/dom/media/webrtc/tests/mochitests/identity/test_setIdentityProviderWithErrors.html
new file mode 100644
index 0000000000..ce6832d1e6
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_setIdentityProviderWithErrors.html
@@ -0,0 +1,57 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript">var scriptRelativePath = "../";</script>
+  <script type="application/javascript" src="../pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+'use strict';
+  createHTML({
+    title: "Identity Provider returning errors is handled correctly",
+    bug: "942367"
+  });
+
+runNetworkTest(function () {
+  var test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  // No IdP for local.
+  // Remote generates a bad assertion, but that only fails to validate
+  test.pcRemote.setIdentityProvider('example.com',
+                                    { protocol: 'idp.js#bad-validate',
+                                      usernameHint: 'nobody' });
+
+  // Save the peerIdentity promises now, since when they reject they are
+  // replaced and we expect them to be rejected this time
+  var peerIdentityLocal = test.pcLocal._pc.peerIdentity;
+  var peerIdentityRemote = test.pcRemote._pc.peerIdentity;
+
+  test.chain.append([
+    function ONLY_REMOTE_SDP_INCLUDES_IDENTITY_ASSERTION(t) {
+      ok(!t.originalOffer.sdp.includes('a=identity'),
+         'a=identity not contained in the offer SDP');
+      ok(t.originalAnswer.sdp.includes('a=identity'),
+         'a=identity is contained in the answer SDP');
+    },
+    function PEER_IDENTITY_IS_EMPTY(t) {
+      // we are only waiting for the local side to complete
+      // an error on the remote side is immediately fatal though
+      return Promise.race([
+        peerIdentityLocal.then(
+          () => ok(false, t.pcLocal + ' incorrectly received valid peer identity'),
+          e => ok(e, t.pcLocal + ' correctly failed to validate peer identity')),
+        peerIdentityRemote.then(
+          () => ok(false, t.pcRemote + ' incorrecly received a valid peer identity'),
+          e => ok(false, t.pcRemote + ' incorrectly rejected peer identity'))
+      ]);
+    }
+  ]);
+
+  return test.run();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/mediaStreamPlayback.js b/dom/media/webrtc/tests/mochitests/mediaStreamPlayback.js
new file mode 100644
index 0000000000..44c1c78ea0
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/mediaStreamPlayback.js
@@ -0,0 +1,241 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+const ENDED_TIMEOUT_LENGTH = 30000;
+
+/* The time we wait depends primarily on the canplaythrough event firing
+ * Note: this needs to be at least 30s because the
+ *       B2G emulator in VMs is really slow. */
+const VERIFYPLAYING_TIMEOUT_LENGTH = 60000;
+
+/**
+ * This class manages playback of a HTMLMediaElement with a MediaStream.
+ * When constructed by a caller, an object instance is created with
+ * a media element and a media stream object.
+ *
+ * @param {HTMLMediaElement} mediaElement the media element for playback
+ * @param {MediaStream} mediaStream the media stream used in
+ *                                  the mediaElement for playback
+ */
+function MediaStreamPlayback(mediaElement, mediaStream) {
+  this.mediaElement = mediaElement;
+  this.mediaStream = mediaStream;
+}
+
+MediaStreamPlayback.prototype = {
+  /**
+   * Starts media element with a media stream, runs it until a canplaythrough
+   * and timeupdate event fires, and calls stop() on all its tracks.
+   *
+   * @param {Boolean} isResume specifies if this media element is being resumed
+   *                           from a previous run
+   */
+  playMedia(isResume) {
+    this.startMedia(isResume);
+    return this.verifyPlaying()
+      .then(() => this.stopTracksForStreamInMediaPlayback())
+      .then(() => this.detachFromMediaElement());
+  },
+
+  /**
+   * Stops the local media stream's tracks while it's currently in playback in
+   * a media element.
+   *
+   * Precondition: The media stream and element should both be actively
+   *               being played. All the stream's tracks must be local.
+   */
+  stopTracksForStreamInMediaPlayback() {
+    var elem = this.mediaElement;
+    return Promise.all([
+      haveEvent(
+        elem,
+        "ended",
+        wait(ENDED_TIMEOUT_LENGTH, new Error("Timeout"))
+      ),
+      ...this.mediaStream
+        .getTracks()
+        .map(t => (t.stop(), haveNoEvent(t, "ended"))),
+    ]);
+  },
+
+  /**
+   * Starts media with a media stream, runs it until a canplaythrough and
+   * timeupdate event fires, and detaches from the element without stopping media.
+   *
+   * @param {Boolean} isResume specifies if this media element is being resumed
+   *                           from a previous run
+   */
+  playMediaWithoutStoppingTracks(isResume) {
+    this.startMedia(isResume);
+    return this.verifyPlaying().then(() => this.detachFromMediaElement());
+  },
+
+  /**
+   * Starts the media with the associated stream.
+   *
+   * @param {Boolean} isResume specifies if the media element playback
+   *                           is being resumed from a previous run
+   */
+  startMedia(isResume) {
+    // If we're playing media element for the first time, check that time is zero.
+    if (!isResume) {
+      is(
+        this.mediaElement.currentTime,
+        0,
+        "Before starting the media element, currentTime = 0"
+      );
+    }
+    this.canPlayThroughFired = listenUntil(
+      this.mediaElement,
+      "canplaythrough",
+      () => true
+    );
+
+    // Hooks up the media stream to the media element and starts playing it
+    this.mediaElement.srcObject = this.mediaStream;
+    this.mediaElement.play();
+  },
+
+  /**
+   * Verifies that media is playing.
+   */
+  verifyPlaying() {
+    var lastElementTime = this.mediaElement.currentTime;
+
+    var mediaTimeProgressed = listenUntil(
+      this.mediaElement,
+      "timeupdate",
+      () => this.mediaElement.currentTime > lastElementTime
+    );
+
+    return timeout(
+      Promise.all([this.canPlayThroughFired, mediaTimeProgressed]),
+      VERIFYPLAYING_TIMEOUT_LENGTH,
+      "verifyPlaying timed out"
+    ).then(() => {
+      is(this.mediaElement.paused, false, "Media element should be playing");
+      is(
+        this.mediaElement.duration,
+        Number.POSITIVE_INFINITY,
+        "Duration should be infinity"
+      );
+
+      // When the media element is playing with a real-time stream, we
+      // constantly switch between having data to play vs. queuing up data,
+      // so we can only check that the ready state is one of those two values
+      ok(
+        this.mediaElement.readyState === HTMLMediaElement.HAVE_ENOUGH_DATA ||
+          this.mediaElement.readyState === HTMLMediaElement.HAVE_CURRENT_DATA,
+        "Ready state shall be HAVE_ENOUGH_DATA or HAVE_CURRENT_DATA"
+      );
+
+      is(this.mediaElement.seekable.length, 0, "Seekable length shall be zero");
+      is(this.mediaElement.buffered.length, 0, "Buffered length shall be zero");
+
+      is(
+        this.mediaElement.seeking,
+        false,
+        "MediaElement is not seekable with MediaStream"
+      );
+      ok(
+        isNaN(this.mediaElement.startOffsetTime),
+        "Start offset time shall not be a number"
+      );
+      is(
+        this.mediaElement.defaultPlaybackRate,
+        1,
+        "DefaultPlaybackRate should be 1"
+      );
+      is(this.mediaElement.playbackRate, 1, "PlaybackRate should be 1");
+      is(this.mediaElement.preload, "none", 'Preload should be "none"');
+      is(this.mediaElement.src, "", "No src should be defined");
+      is(
+        this.mediaElement.currentSrc,
+        "",
+        "Current src should still be an empty string"
+      );
+    });
+  },
+
+  /**
+   * Detaches from the element without stopping the media.
+   *
+   * Precondition: The media stream and element should both be actively
+   *               being played.
+   */
+  detachFromMediaElement() {
+    this.mediaElement.pause();
+    this.mediaElement.srcObject = null;
+  },
+};
+
+// haxx to prevent SimpleTest from failing at window.onload
+function addLoadEvent() {}
+
+/* import-globals-from /testing/mochitest/tests/SimpleTest/SimpleTest.js */
+/* import-globals-from head.js */
+const scriptsReady = Promise.all(
+  ["/tests/SimpleTest/SimpleTest.js", "head.js"].map(script => {
+    const el = document.createElement("script");
+    el.src = script;
+    document.head.appendChild(el);
+    return new Promise(r => (el.onload = r));
+  })
+);
+
+function createHTML(options) {
+  return scriptsReady.then(() => realCreateHTML(options));
+}
+
+async function runTest(testFunction) {
+  await Promise.all([
+    scriptsReady,
+    SpecialPowers.pushPrefEnv({
+      set: [["media.navigator.permission.fake", true]],
+    }),
+  ]);
+  await runTestWhenReady(async (...args) => {
+    await testFunction(...args);
+    await noGum();
+  });
+}
+
+// noGum - Helper to detect whether active guM tracks still exist.
+//
+// Note it relies on the permissions system to detect active tracks, so it won't
+// catch getUserMedia use while media.navigator.permission.disabled is true
+// (which is common in automation), UNLESS we set
+// media.navigator.permission.fake to true also, like runTest() does above.
+async function noGum() {
+  if (!navigator.mediaDevices) {
+    // No mediaDevices, then gUM cannot have been called either.
+    return;
+  }
+  const mediaManagerService = Cc[
+    "@mozilla.org/mediaManagerService;1"
+  ].getService(Ci.nsIMediaManagerService);
+
+  const hasCamera = {};
+  const hasMicrophone = {};
+  mediaManagerService.mediaCaptureWindowState(
+    window,
+    hasCamera,
+    hasMicrophone,
+    {},
+    {},
+    {},
+    {},
+    false
+  );
+  is(
+    hasCamera.value,
+    mediaManagerService.STATE_NOCAPTURE,
+    "Test must leave no active camera gUM tracks behind."
+  );
+  is(
+    hasMicrophone.value,
+    mediaManagerService.STATE_NOCAPTURE,
+    "Test must leave no active microphone gUM tracks behind."
+  );
+}
diff --git a/dom/media/webrtc/tests/mochitests/mochitest.ini b/dom/media/webrtc/tests/mochitests/mochitest.ini
new file mode 100644
index 0000000000..d1a9800984
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/mochitest.ini
@@ -0,0 +1,65 @@
+[DEFAULT]
+tags = mtg webrtc
+subsuite = media
+scheme = https
+support-files =
+  head.js
+  dataChannel.js
+  mediaStreamPlayback.js
+  network.js
+  nonTrickleIce.js
+  pc.js
+  stats.js
+  templates.js
+  test_enumerateDevices_iframe.html
+  test_enumerateDevices_iframe_pre_gum.html
+  test_getUserMedia_permission_iframe.html
+  NetworkPreparationChromeScript.js
+  blacksilence.js
+  turnConfig.js
+  sdpUtils.js
+  addTurnsSelfsignedCert.js
+  parser_rtp.js
+  peerconnection_audio_forced_sample_rate.js
+  iceTestUtils.js
+  simulcast.js
+  helpers_from_wpt/sdp.js
+  !/dom/canvas/test/captureStream_common.js
+  !/dom/canvas/test/webgl-mochitest/webgl-util.js
+  !/dom/media/test/manifest.js
+  !/dom/media/test/seek.webm
+  !/dom/media/test/gizmo.mp4
+  !/docshell/test/navigation/blank.html
+prefs =
+  focusmanager.testmode=true # emulate focus
+  privacy.partition.network_state=false
+  network.proxy.allow_hijacking_localhost=true
+  media.devices.enumerate.legacy.enabled=false
+
+[test_1488832.html]
+skip-if =
+  os == 'linux' # Bug 1714410
+[test_1717318.html]
+[test_a_noOp.html]
+scheme=http
+[test_enumerateDevices.html]
+[test_enumerateDevices_getUserMediaFake.html]
+[test_enumerateDevices_legacy.html]
+[test_enumerateDevices_navigation.html]
+skip-if = true # Disabled because it is a racy test and causes timeouts, see bug 1650932
+[test_fingerprinting_resistance.html]
+skip-if =
+  os == "linux" && asan # Bug 1646309 - low frequency intermittent
+[test_forceSampleRate.html]
+scheme=http
+[test_groupId.html]
+[test_multi_mics.html]
+skip-if = os == 'android'
+[test_ondevicechange.html]
+run-sequentially = sets prefs that may disrupt other tests
+[test_setSinkId.html]
+skip-if =
+  os != 'linux' # the only platform with real devices
+[test_setSinkId_default_addTrack.html]
+[test_setSinkId_preMutedElement.html]
+[test_unfocused_pref.html]
diff --git a/dom/media/webrtc/tests/mochitests/mochitest_datachannel.ini b/dom/media/webrtc/tests/mochitests/mochitest_datachannel.ini
new file mode 100644
index 0000000000..881421af0e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/mochitest_datachannel.ini
@@ -0,0 +1,52 @@
+[DEFAULT]
+tags = mtg webrtc
+subsuite = media
+scheme = https
+support-files =
+  head.js
+  dataChannel.js
+  mediaStreamPlayback.js
+  network.js
+  nonTrickleIce.js
+  pc.js
+  stats.js
+  templates.js
+  test_enumerateDevices_iframe.html
+  test_getUserMedia_permission_iframe.html
+  NetworkPreparationChromeScript.js
+  blacksilence.js
+  turnConfig.js
+  sdpUtils.js
+  addTurnsSelfsignedCert.js
+  parser_rtp.js
+  peerconnection_audio_forced_sample_rate.js
+  iceTestUtils.js
+  simulcast.js
+  helpers_from_wpt/sdp.js
+  !/dom/canvas/test/captureStream_common.js
+  !/dom/canvas/test/webgl-mochitest/webgl-util.js
+  !/dom/media/test/manifest.js
+  !/dom/media/test/seek.webm
+  !/dom/media/test/gizmo.mp4
+  !/docshell/test/navigation/blank.html
+prefs =
+  focusmanager.testmode=true # emulate focus
+  privacy.partition.network_state=false
+  network.proxy.allow_hijacking_localhost=true
+
+[test_dataChannel_basicAudio.html]
+[test_dataChannel_basicAudioVideo.html]
+[test_dataChannel_basicAudioVideoCombined.html]
+[test_dataChannel_basicAudioVideoNoBundle.html]
+[test_dataChannel_basicDataOnly.html]
+[test_dataChannel_basicVideo.html]
+[test_dataChannel_bug1013809.html]
+[test_dataChannel_dataOnlyBufferedAmountLow.html]
+scheme=http
+[test_dataChannel_dtlsVersions.html]
+[test_dataChannel_hostnameObfuscation.html]
+scheme=http
+[test_dataChannel_noOffer.html]
+scheme=http
+[test_dataChannel_stats.html]
+scheme=http
diff --git a/dom/media/webrtc/tests/mochitests/mochitest_getusermedia.ini b/dom/media/webrtc/tests/mochitests/mochitest_getusermedia.ini
new file mode 100644
index 0000000000..02c8272b5b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/mochitest_getusermedia.ini
@@ -0,0 +1,105 @@
+[DEFAULT]
+tags = mtg webrtc
+subsuite = media
+scheme = https
+support-files =
+  head.js
+  dataChannel.js
+  mediaStreamPlayback.js
+  network.js
+  nonTrickleIce.js
+  pc.js
+  stats.js
+  templates.js
+  test_enumerateDevices_iframe.html
+  test_getUserMedia_permission_iframe.html
+  NetworkPreparationChromeScript.js
+  blacksilence.js
+  turnConfig.js
+  sdpUtils.js
+  addTurnsSelfsignedCert.js
+  parser_rtp.js
+  peerconnection_audio_forced_sample_rate.js
+  iceTestUtils.js
+  simulcast.js
+  helpers_from_wpt/sdp.js
+  !/dom/canvas/test/captureStream_common.js
+  !/dom/canvas/test/webgl-mochitest/webgl-util.js
+  !/dom/media/test/manifest.js
+  !/dom/media/test/seek.webm
+  !/dom/media/test/gizmo.mp4
+  !/docshell/test/navigation/blank.html
+prefs =
+  focusmanager.testmode=true # emulate focus
+  privacy.partition.network_state=false
+  network.proxy.allow_hijacking_localhost=true
+  media.devices.enumerate.legacy.enabled=false
+
+[test_defaultAudioConstraints.html]
+skip-if = os == 'mac'
+  os == 'win'
+  toolkit == 'android' # Bug 1404995, no loopback devices on some platforms
+[test_getUserMedia_GC_MediaStream.html]
+[test_getUserMedia_active_autoplay.html]
+[test_getUserMedia_addTrackRemoveTrack.html]
+[test_getUserMedia_addtrack_removetrack_events.html]
+[test_getUserMedia_audioCapture.html]
+skip-if = toolkit == 'android'
+  (os == "win" && processor == "aarch64") # android(Bug 1189784, timeouts on 4.3 emulator), android(Bug 1264333), aarch64 due to 1538359
+[test_getUserMedia_audioConstraints.html]
+skip-if = os == 'mac'
+  os == 'win'
+  toolkit == 'android' # Bug 1404995, no loopback devices on some platforms
+[test_getUserMedia_audioConstraints_concurrentIframes.html]
+skip-if = os == 'mac'
+  os == 'win'
+  toolkit == 'android'
+  (os == 'linux' && debug)  # Bug 1404995, no loopback devices on some platforms # Bug 1481101
+  os == "linux" && !debug && !fission # bug 1645930, lower frequency intermittent
+[test_getUserMedia_audioConstraints_concurrentStreams.html]
+skip-if = os == 'mac'
+  os == 'win'
+  toolkit == 'android' # Bug 1404995, no loopback devices on some platforms
+[test_getUserMedia_basicAudio.html]
+[test_getUserMedia_basicAudio_loopback.html]
+skip-if = os == 'mac'
+  os == 'win'
+  toolkit == 'android' # Bug 1404995, no loopback devices on some platforms
+[test_getUserMedia_basicScreenshare.html]
+skip-if =
+  toolkit == 'android'     # no screenshare on android
+  apple_silicon            # bug 1707742
+[test_getUserMedia_basicTabshare.html]
+skip-if = 
+  toolkit == 'android' # no windowshare on android
+[test_getUserMedia_basicVideo.html]
+[test_getUserMedia_basicVideoAudio.html]
+[test_getUserMedia_basicVideo_playAfterLoadedmetadata.html]
+[test_getUserMedia_basicWindowshare.html]
+skip-if = toolkit == 'android' # no windowshare on android
+[test_getUserMedia_bug1223696.html]
+[test_getUserMedia_callbacks.html]
+[test_getUserMedia_constraints.html]
+[test_getUserMedia_cubebDisabled.html]
+[test_getUserMedia_cubebDisabledFakeStreams.html]
+[test_getUserMedia_getTrackById.html]
+[test_getUserMedia_gumWithinGum.html]
+[test_getUserMedia_loadedmetadata.html]
+[test_getUserMedia_mediaElementCapture_audio.html]
+[test_getUserMedia_mediaElementCapture_tracks.html]
+[test_getUserMedia_mediaElementCapture_video.html]
+[test_getUserMedia_mediaStreamClone.html]
+[test_getUserMedia_mediaStreamConstructors.html]
+[test_getUserMedia_mediaStreamTrackClone.html]
+[test_getUserMedia_nonDefaultRate.html]
+[test_getUserMedia_peerIdentity.html]
+[test_getUserMedia_permission.html]
+[test_getUserMedia_playAudioTwice.html]
+[test_getUserMedia_playVideoAudioTwice.html]
+[test_getUserMedia_playVideoTwice.html]
+[test_getUserMedia_scarySources.html]
+skip-if = toolkit == 'android' # no screenshare or windowshare on android
+[test_getUserMedia_spinEventLoop.html]
+[test_getUserMedia_trackCloneCleanup.html]
+[test_getUserMedia_trackEnded.html]
+
diff --git a/dom/media/webrtc/tests/mochitests/mochitest_peerconnection.ini b/dom/media/webrtc/tests/mochitests/mochitest_peerconnection.ini
new file mode 100644
index 0000000000..37fb551435
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/mochitest_peerconnection.ini
@@ -0,0 +1,311 @@
+[DEFAULT]
+tags = mtg webrtc
+subsuite = media
+scheme = https
+support-files =
+  head.js
+  dataChannel.js
+  mediaStreamPlayback.js
+  network.js
+  nonTrickleIce.js
+  pc.js
+  stats.js
+  templates.js
+  test_enumerateDevices_iframe.html
+  test_getUserMedia_permission_iframe.html
+  NetworkPreparationChromeScript.js
+  blacksilence.js
+  turnConfig.js
+  sdpUtils.js
+  addTurnsSelfsignedCert.js
+  parser_rtp.js
+  peerconnection_audio_forced_sample_rate.js
+  iceTestUtils.js
+  simulcast.js
+  helpers_from_wpt/sdp.js
+  !/dom/canvas/test/captureStream_common.js
+  !/dom/canvas/test/webgl-mochitest/webgl-util.js
+  !/dom/media/test/manifest.js
+  !/dom/media/test/seek.webm
+  !/dom/media/test/gizmo.mp4
+  !/docshell/test/navigation/blank.html
+prefs =
+  focusmanager.testmode=true # emulate focus
+  privacy.partition.network_state=false
+  network.proxy.allow_hijacking_localhost=true
+  media.devices.enumerate.legacy.enabled=false
+
+[test_peerConnection_addAudioTrackToExistingVideoStream.html]
+[test_peerConnection_addDataChannel.html]
+[test_peerConnection_addDataChannelNoBundle.html]
+[test_peerConnection_addSecondAudioStream.html]
+[test_peerConnection_addSecondAudioStreamNoBundle.html]
+[test_peerConnection_addSecondVideoStream.html]
+[test_peerConnection_addSecondVideoStreamNoBundle.html]
+[test_peerConnection_addtrack_removetrack_events.html]
+[test_peerConnection_answererAddSecondAudioStream.html]
+[test_peerConnection_audioChannels.html]
+[test_peerConnection_audioCodecs.html]
+[test_peerConnection_audioContributingSources.html]
+[test_peerConnection_audioRenegotiationInactiveAnswer.html]
+[test_peerConnection_audioSynchronizationSources.html]
+[test_peerConnection_audioSynchronizationSourcesUnidirectional.html]
+[test_peerConnection_basicAudio.html]
+[test_peerConnection_basicAudioDynamicPtMissingRtpmap.html]
+[test_peerConnection_basicAudioNATRelay.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+scheme=http
+[test_peerConnection_basicAudioNATRelayTCP.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioNATRelayTCPWithStun300.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioNATRelayTLS.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioNATRelayWithStun300.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioNATSrflx.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioNoisyUDPBlock.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioPcmaPcmuOnly.html]
+[test_peerConnection_basicAudioRelayPolicy.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioRequireEOC.html]
+[test_peerConnection_basicAudioVerifyRtpHeaderExtensions.html]
+[test_peerConnection_basicAudioVideo.html]
+[test_peerConnection_basicAudioVideoCombined.html]
+[test_peerConnection_basicAudioVideoNoBundle.html]
+[test_peerConnection_basicAudioVideoNoBundleNoRtcpMux.html]
+[test_peerConnection_basicAudioVideoNoRtcpMux.html]
+[test_peerConnection_basicAudioVideoTransceivers.html]
+[test_peerConnection_basicAudioVideoVerifyExtmap.html]
+[test_peerConnection_basicAudioVideoVerifyExtmapSendonly.html]
+[test_peerConnection_basicAudioVideoVerifyTooLongMidFails.html]
+[test_peerConnection_basicAudio_forced_higher_rate.html]
+[test_peerConnection_basicAudio_forced_lower_rate.html]
+[test_peerConnection_basicH264Video.html]
+skip-if =
+    toolkit == 'android' && is_emulator  # Bug 1355786, No h264 support on android emulator
+[test_peerConnection_basicScreenshare.html]
+skip-if = toolkit == 'android' # no screenshare on android
+[test_peerConnection_basicVideo.html]
+[test_peerConnection_basicVideoVerifyRtpHeaderExtensions.html]
+[test_peerConnection_basicWindowshare.html]
+skip-if = toolkit == 'android' # no screenshare on android
+[test_peerConnection_bug1013809.html]
+[test_peerConnection_bug1042791.html]
+skip-if = (toolkit == 'android' && is_emulator) # Bug 1355786, No h264 support on android emulator
+[test_peerConnection_bug1227781.html]
+scheme=http
+[test_peerConnection_bug1512281.html]
+fail-if = 1
+[test_peerConnection_bug1773067.html]
+[test_peerConnection_bug822674.html]
+scheme=http
+[test_peerConnection_bug825703.html]
+scheme=http
+[test_peerConnection_bug827843.html]
+[test_peerConnection_bug834153.html]
+scheme=http
+[test_peerConnection_callbacks.html]
+[test_peerConnection_captureStream_canvas_2d.html]
+scheme=http
+[test_peerConnection_captureStream_canvas_2d_noSSRC.html]
+scheme=http
+[test_peerConnection_captureStream_canvas_webgl.html]
+scheme=http
+[test_peerConnection_capturedVideo.html]
+tags=capturestream
+skip-if = toolkit == 'android' # android(Bug 1189784, timeouts on 4.3 emulator), Bug 1264340
+[test_peerConnection_certificates.html]
+[test_peerConnection_checkPacketDumpHook.html]
+[test_peerConnection_close.html]
+scheme=http
+[test_peerConnection_closeDuringIce.html]
+[test_peerConnection_codecNegotiationFailure.html]
+[test_peerConnection_constructedStream.html]
+[test_peerConnection_disabledVideoPreNegotiation.html]
+[test_peerConnection_encodingsNegotiation.html]
+[test_peerConnection_errorCallbacks.html]
+scheme=http
+[test_peerConnection_extmapRenegotiation.html]
+[test_peerConnection_forwarding_basicAudioVideoCombined.html]
+skip-if = toolkit == 'android'  # Bug 1189784
+[test_peerConnection_gatherWithSetConfiguration.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_gatherWithStun300.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_gatherWithStun300IPv6.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    os == 'mac' # no ipv6 support on OS X testers (bug 1710706)
+    os == 'win' # no ipv6 support on windows testers (bug 1710706)
+scheme=http
+[test_peerConnection_glean.html]
+[test_peerConnection_iceFailure.html]
+skip-if = true # (Bug 1180388 for win, mac and linux), android(Bug 1189784), Bug 1180388
+scheme=http
+[test_peerConnection_insertDTMF.html]
+[test_peerConnection_localReofferRollback.html]
+[test_peerConnection_localRollback.html]
+[test_peerConnection_maxFsConstraint.html]
+[test_peerConnection_multiple_captureStream_canvas_2d.html]
+scheme=http
+[test_peerConnection_noTrickleAnswer.html]
+[test_peerConnection_noTrickleOffer.html]
+[test_peerConnection_noTrickleOfferAnswer.html]
+[test_peerConnection_nonDefaultRate.html]
+[test_peerConnection_offerRequiresReceiveAudio.html]
+[test_peerConnection_offerRequiresReceiveVideo.html]
+[test_peerConnection_offerRequiresReceiveVideoAudio.html]
+[test_peerConnection_portRestrictions.html]
+[test_peerConnection_promiseSendOnly.html]
+[test_peerConnection_recordReceiveTrack.html]
+[test_peerConnection_relayOnly.html]
+disabled=bug 1612063 # test is racy
+[test_peerConnection_remoteReofferRollback.html]
+[test_peerConnection_remoteRollback.html]
+[test_peerConnection_removeAudioTrack.html]
+[test_peerConnection_removeThenAddAudioTrack.html]
+[test_peerConnection_removeThenAddAudioTrackNoBundle.html]
+[test_peerConnection_removeThenAddVideoTrack.html]
+[test_peerConnection_removeThenAddVideoTrackNoBundle.html]
+[test_peerConnection_removeVideoTrack.html]
+[test_peerConnection_renderAfterRenegotiation.html]
+scheme=http
+[test_peerConnection_replaceNullTrackThenRenegotiateAudio.html]
+[test_peerConnection_replaceNullTrackThenRenegotiateVideo.html]
+[test_peerConnection_replaceTrack.html]
+[test_peerConnection_replaceTrack_camera.html]
+skip-if = toolkit == 'android' # Bug 1614460
+[test_peerConnection_replaceTrack_disabled.html]
+skip-if =
+  toolkit == 'android'        # Bug 1614460
+[test_peerConnection_replaceTrack_microphone.html]
+[test_peerConnection_replaceVideoThenRenegotiate.html]
+[test_peerConnection_restartIce.html]
+[test_peerConnection_restartIceBadAnswer.html]
+[test_peerConnection_restartIceLocalAndRemoteRollback.html]
+[test_peerConnection_restartIceLocalAndRemoteRollbackNoSubsequentRestart.html]
+[test_peerConnection_restartIceLocalRollback.html]
+[test_peerConnection_restartIceLocalRollbackNoSubsequentRestart.html]
+[test_peerConnection_restartIceNoBundle.html]
+[test_peerConnection_restartIceNoBundleNoRtcpMux.html]
+[test_peerConnection_restartIceNoRtcpMux.html]
+[test_peerConnection_restrictBandwidthTargetBitrate.html]
+[test_peerConnection_restrictBandwidthWithTias.html]
+[test_peerConnection_rtcp_rsize.html]
+[test_peerConnection_scaleResolution.html]
+[test_peerConnection_scaleResolution_oldSetParameters.html]
+[test_peerConnection_sender_and_receiver_stats.html]
+[test_peerConnection_setLocalAnswerInHaveLocalOffer.html]
+[test_peerConnection_setLocalAnswerInStable.html]
+[test_peerConnection_setLocalOfferInHaveRemoteOffer.html]
+[test_peerConnection_setParameters.html]
+[test_peerConnection_setParameters_maxFramerate.html]
+[test_peerConnection_setParameters_maxFramerate_oldSetParameters.html]
+[test_peerConnection_setParameters_oldSetParameters.html]
+[test_peerConnection_setParameters_scaleResolutionDownBy.html]
+skip-if = (os == 'win' && processor == 'aarch64') # aarch64 due to bug 1537567
+[test_peerConnection_setParameters_scaleResolutionDownBy_oldSetParameters.html]
+skip-if = (os == 'win' && processor == 'aarch64') # aarch64 due to bug 1537567
+[test_peerConnection_setRemoteAnswerInHaveRemoteOffer.html]
+[test_peerConnection_setRemoteAnswerInStable.html]
+[test_peerConnection_setRemoteOfferInHaveLocalOffer.html]
+[test_peerConnection_simulcastAnswer.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastAnswer_lowResFirst.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastAnswer_lowResFirst_oldSetParameters.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastAnswer_oldSetParameters.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastOddResolution.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastOddResolution_oldSetParameters.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastOffer.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastOffer_lowResFirst.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastOffer_lowResFirst_oldSetParameters.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastOffer_oldSetParameters.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_stats.html]
+[test_peerConnection_stats_jitter.html]
+skip-if = tsan # Bug 1672590, TSan is just too slow to pass this test
+[test_peerConnection_stats_oneway.html]
+[test_peerConnection_stats_relayProtocol.html]
+skip-if =
+    toolkit == 'android' # android(Bug 1189784, timeouts on 4.3 emulator, Bug 1373858, Bug 1521117)
+    socketprocess_e10s
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_stereoFmtpPref.html]
+[test_peerConnection_syncSetDescription.html]
+[test_peerConnection_telephoneEventFirst.html]
+[test_peerConnection_threeUnbundledConnections.html]
+[test_peerConnection_throwInCallbacks.html]
+[test_peerConnection_toJSON.html]
+scheme=http
+[test_peerConnection_trackDisabling.html]
+skip-if = toolkit == 'android' # Bug 1614460
+[test_peerConnection_trackDisabling_clones.html]
+[test_peerConnection_trackless_sender_stats.html]
+[test_peerConnection_twoAudioStreams.html]
+[test_peerConnection_twoAudioTracksInOneStream.html]
+[test_peerConnection_twoAudioVideoStreams.html]
+[test_peerConnection_twoAudioVideoStreamsCombined.html]
+skip-if = (toolkit == 'android') || (os == 'linux' && asan) # android(Bug 1189784), Bug 1480942 for Linux asan
+[test_peerConnection_twoAudioVideoStreamsCombinedNoBundle.html]
+skip-if =
+  (toolkit == 'android') # Bug 1189784
+  (os == 'linux' && asan) # Bug 1480942
+  (os == 'win' && processor == 'aarch64') # Bug 1777081
+[test_peerConnection_twoVideoStreams.html]
+[test_peerConnection_twoVideoTracksInOneStream.html]
+[test_peerConnection_verifyAudioAfterRenegotiation.html]
+skip-if =
+  os == "android" && processor == "x86_64" && !debug # Bug 1783287
+[test_peerConnection_verifyDescriptions.html]
+[test_peerConnection_verifyVideoAfterRenegotiation.html]
+[test_peerConnection_videoCodecs.html]
+skip-if =
+  toolkit == 'android' # android(Bug 1614460)
+  win10_2004 && !debug # Bug 1777082
+[test_peerConnection_videoRenegotiationInactiveAnswer.html]
+[test_peerConnection_webAudio.html]
+tags = webaudio webrtc
+scheme=http
+[test_selftest.html]
+# Bug 1227781: Crash with bogus TURN server.
+scheme=http
diff --git a/dom/media/webrtc/tests/mochitests/network.js b/dom/media/webrtc/tests/mochitests/network.js
new file mode 100644
index 0000000000..223721b111
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/network.js
@@ -0,0 +1,16 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+"use strict";
+
+/**
+ * A stub function for preparing the network if needed
+ *
+ */
+async function startNetworkAndTest() {}
+
+/**
+ * A stub function to shutdown the network if needed
+ */
+async function networkTestFinished() {}
diff --git a/dom/media/webrtc/tests/mochitests/nonTrickleIce.js b/dom/media/webrtc/tests/mochitests/nonTrickleIce.js
new file mode 100644
index 0000000000..9361944791
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/nonTrickleIce.js
@@ -0,0 +1,97 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+function removeTrickleOption(desc) {
+  var sdp = desc.sdp.replace(/\r\na=ice-options:trickle\r\n/, "\r\n");
+  return new RTCSessionDescription({ type: desc.type, sdp });
+}
+
+function makeOffererNonTrickle(chain) {
+  chain.replace("PC_LOCAL_SETUP_ICE_HANDLER", [
+    function PC_LOCAL_SETUP_NOTRICKLE_ICE_HANDLER(test) {
+      // We need to install this callback before calling setLocalDescription
+      // otherwise we might miss callbacks
+      test.pcLocal.setupIceCandidateHandler(test, () => {});
+      // We ignore ICE candidates because we want the full offer
+    },
+  ]);
+  chain.replace("PC_REMOTE_GET_OFFER", [
+    function PC_REMOTE_GET_FULL_OFFER(test) {
+      return test.pcLocal.endOfTrickleIce.then(() => {
+        test._local_offer = removeTrickleOption(test.pcLocal.localDescription);
+        test._offer_constraints = test.pcLocal.constraints;
+        test._offer_options = test.pcLocal.offerOptions;
+      });
+    },
+  ]);
+  chain.insertAfter("PC_REMOTE_SANE_REMOTE_SDP", [
+    function PC_REMOTE_REQUIRE_REMOTE_SDP_CANDIDATES(test) {
+      info(
+        "test.pcLocal.localDescription.sdp: " +
+          JSON.stringify(test.pcLocal.localDescription.sdp)
+      );
+      info("test._local_offer.sdp" + JSON.stringify(test._local_offer.sdp));
+      is(
+        test.pcRemote._pc.canTrickleIceCandidates,
+        false,
+        "Remote thinks that trickle isn't supported"
+      );
+      ok(!test.localRequiresTrickleIce, "Local does NOT require trickle");
+      ok(
+        test._local_offer.sdp.includes("a=candidate"),
+        "offer has ICE candidates"
+      );
+      ok(
+        test._local_offer.sdp.includes("a=end-of-candidates"),
+        "offer has end-of-candidates"
+      );
+    },
+  ]);
+  chain.remove("PC_REMOTE_CHECK_CAN_TRICKLE_SYNC");
+}
+
+function makeAnswererNonTrickle(chain) {
+  chain.replace("PC_REMOTE_SETUP_ICE_HANDLER", [
+    function PC_REMOTE_SETUP_NOTRICKLE_ICE_HANDLER(test) {
+      // We need to install this callback before calling setLocalDescription
+      // otherwise we might miss callbacks
+      test.pcRemote.setupIceCandidateHandler(test, () => {});
+      // We ignore ICE candidates because we want the full offer
+    },
+  ]);
+  chain.replace("PC_LOCAL_GET_ANSWER", [
+    function PC_LOCAL_GET_FULL_ANSWER(test) {
+      return test.pcRemote.endOfTrickleIce.then(() => {
+        test._remote_answer = removeTrickleOption(
+          test.pcRemote.localDescription
+        );
+        test._answer_constraints = test.pcRemote.constraints;
+      });
+    },
+  ]);
+  chain.insertAfter("PC_LOCAL_SANE_REMOTE_SDP", [
+    function PC_LOCAL_REQUIRE_REMOTE_SDP_CANDIDATES(test) {
+      info(
+        "test.pcRemote.localDescription.sdp: " +
+          JSON.stringify(test.pcRemote.localDescription.sdp)
+      );
+      info("test._remote_answer.sdp" + JSON.stringify(test._remote_answer.sdp));
+      is(
+        test.pcLocal._pc.canTrickleIceCandidates,
+        false,
+        "Local thinks that trickle isn't supported"
+      );
+      ok(!test.remoteRequiresTrickleIce, "Remote does NOT require trickle");
+      ok(
+        test._remote_answer.sdp.includes("a=candidate"),
+        "answer has ICE candidates"
+      );
+      ok(
+        test._remote_answer.sdp.includes("a=end-of-candidates"),
+        "answer has end-of-candidates"
+      );
+    },
+  ]);
+  chain.remove("PC_LOCAL_CHECK_CAN_TRICKLE_SYNC");
+}
diff --git a/dom/media/webrtc/tests/mochitests/parser_rtp.js b/dom/media/webrtc/tests/mochitests/parser_rtp.js
new file mode 100644
index 0000000000..2275c1f787
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/parser_rtp.js
@@ -0,0 +1,131 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+"use strict";
+
+/*
+ * Parses an RTP packet
+ * @param buffer an ArrayBuffer that contains the packet
+ * @return { type: "rtp", header: {...}, payload: a DataView }
+ */
+var ParseRtpPacket = buffer => {
+  // DataView.getFooInt returns big endian numbers by default
+  let view = new DataView(buffer);
+
+  // Standard Header Fields
+  // https://tools.ietf.org/html/rfc3550#section-5.1
+  //  0                   1                   2                   3
+  //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |V=2|P|X|  CC   |M|     PT      |       sequence number         |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |                           timestamp                           |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |           synchronization source (SSRC) identifier            |
+  // +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+  // |            contributing source (CSRC) identifiers             |
+  // |                             ....                              |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+  let header = {};
+  let offset = 0;
+  // Note that incrementing the offset happens as close to reading the data as
+  // possible. This simplifies ensuring that the number of read bytes and the
+  // offset increment match. Data may be manipulated between when the offset is
+  // incremented and before the next read.
+  let byte = view.getUint8(offset);
+  offset++;
+  // Version            2 Bit
+  header.version = (0xc0 & byte) >> 6;
+  // Padding            1 Bit
+  header.padding = (0x30 & byte) >> 5;
+  // Extension          1 Bit
+  header.extensionsPresent = (0x10 & byte) >> 4 == 1;
+  // CSRC count         4 Bit
+  header.csrcCount = 0xf & byte;
+
+  byte = view.getUint8(offset);
+  offset++;
+  // Marker             1 Bit
+  header.marker = (0x80 & byte) >> 7;
+  // Payload Type       7 Bit
+  header.payloadType = 0x7f & byte;
+  // Sequence Number   16 Bit
+  header.sequenceNumber = view.getUint16(offset);
+  offset += 2;
+  // Timestamp         32 Bit
+  header.timestamp = view.getUint32(offset);
+  offset += 4;
+  // SSRC              32 Bit
+  header.ssrc = view.getUint32(offset);
+  offset += 4;
+
+  // CSRC              32 Bit
+  header.csrcs = [];
+  for (let c = 0; c < header.csrcCount; c++) {
+    header.csrcs.push(view.getUint32(offset));
+    offset += 4;
+  }
+
+  // Extensions
+  header.extensions = [];
+  header.extensionPaddingBytes = 0;
+  header.extensionsTotalLength = 0;
+  if (header.extensionsPresent) {
+    // https://tools.ietf.org/html/rfc3550#section-5.3.1
+    //  0                   1                   2                   3
+    //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |      defined by profile       |           length              |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |                        header extension                       |
+    // |                             ....                              |
+    let addExtension = (id, len) =>
+      header.extensions.push({
+        id,
+        data: new DataView(buffer, offset, len),
+      });
+    let extensionId = view.getUint16(offset);
+    offset += 2;
+    // len is in 32 bit units, not bytes
+    header.extensionsTotalLength = view.getUint16(offset) * 4;
+    offset += 2;
+    // Check for https://tools.ietf.org/html/rfc5285
+    if (extensionId != 0xbede) {
+      // No rfc5285
+      addExtension(extensionId, header.extensionsTotalLength);
+      offset += header.extensionsTotalLength;
+    } else {
+      let expectedEnd = offset + header.extensionsTotalLength;
+      while (offset < expectedEnd) {
+        // We only support "one-byte" extension headers ATM
+        // https://tools.ietf.org/html/rfc5285#section-4.2
+        //  0
+        //  0 1 2 3 4 5 6 7
+        // +-+-+-+-+-+-+-+-+
+        // |  ID   |  len  |
+        // +-+-+-+-+-+-+-+-+
+        byte = view.getUint8(offset);
+        offset++;
+        // Check for padding which can occur between extensions or at the end
+        if (byte == 0) {
+          header.extensionPaddingBytes++;
+          continue;
+        }
+        let id = (byte & 0xf0) >> 4;
+        // Check for the FORBIDDEN id (15), dun dun dun
+        if (id == 15) {
+          // Ignore bytes until until the end of extensions
+          offset = expectedEnd;
+          break;
+        }
+        // the length of the extention is len + 1
+        let len = (byte & 0x0f) + 1;
+        addExtension(id, len);
+        offset += len;
+      }
+    }
+  }
+  return { type: "rtp", header, payload: new DataView(buffer, offset) };
+};
diff --git a/dom/media/webrtc/tests/mochitests/pc.js b/dom/media/webrtc/tests/mochitests/pc.js
new file mode 100644
index 0000000000..36a923fbed
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/pc.js
@@ -0,0 +1,2495 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+"use strict";
+
+const LOOPBACK_ADDR = "127.0.0.";
+
+const iceStateTransitions = {
+  new: ["checking", "closed"], //Note: 'failed' might need to added here
+  //      even though it is not in the standard
+  checking: ["new", "connected", "failed", "closed"], //Note: do we need to
+  // allow 'completed' in
+  // here as well?
+  connected: ["new", "completed", "disconnected", "closed"],
+  completed: ["new", "disconnected", "closed"],
+  disconnected: ["new", "connected", "completed", "failed", "closed"],
+  failed: ["new", "disconnected", "closed"],
+  closed: [],
+};
+
+const signalingStateTransitions = {
+  stable: ["have-local-offer", "have-remote-offer", "closed"],
+  "have-local-offer": [
+    "have-remote-pranswer",
+    "stable",
+    "closed",
+    "have-local-offer",
+  ],
+  "have-remote-pranswer": ["stable", "closed", "have-remote-pranswer"],
+  "have-remote-offer": [
+    "have-local-pranswer",
+    "stable",
+    "closed",
+    "have-remote-offer",
+  ],
+  "have-local-pranswer": ["stable", "closed", "have-local-pranswer"],
+  closed: [],
+};
+
+var makeDefaultCommands = () => {
+  return [].concat(
+    commandsPeerConnectionInitial,
+    commandsGetUserMedia,
+    commandsPeerConnectionOfferAnswer
+  );
+};
+
+/**
+ * This class handles tests for peer connections.
+ *
+ * @constructor
+ * @param {object} [options={}]
+ *        Optional options for the peer connection test
+ * @param {object} [options.commands=commandsPeerConnection]
+ *        Commands to run for the test
+ * @param {bool}   [options.is_local=true]
+ *        true if this test should run the tests for the "local" side.
+ * @param {bool}   [options.is_remote=true]
+ *        true if this test should run the tests for the "remote" side.
+ * @param {object} [options.config_local=undefined]
+ *        Configuration for the local peer connection instance
+ * @param {object} [options.config_remote=undefined]
+ *        Configuration for the remote peer connection instance. If not defined
+ *        the configuration from the local instance will be used
+ */
+function PeerConnectionTest(options) {
+  // If no options are specified make it an empty object
+  options = options || {};
+  options.commands = options.commands || makeDefaultCommands();
+  options.is_local = "is_local" in options ? options.is_local : true;
+  options.is_remote = "is_remote" in options ? options.is_remote : true;
+
+  options.h264 = "h264" in options ? options.h264 : false;
+  options.bundle = "bundle" in options ? options.bundle : true;
+  options.rtcpmux = "rtcpmux" in options ? options.rtcpmux : true;
+  options.opus = "opus" in options ? options.opus : true;
+  options.ssrc = "ssrc" in options ? options.ssrc : true;
+
+  options.config_local = options.config_local || {};
+  options.config_remote = options.config_remote || {};
+
+  if (!options.bundle) {
+    // Make sure neither end tries to use bundle-only!
+    options.config_local.bundlePolicy = "max-compat";
+    options.config_remote.bundlePolicy = "max-compat";
+  }
+
+  if (iceServersArray.length) {
+    if (!options.turn_disabled_local && !options.config_local.iceServers) {
+      options.config_local.iceServers = iceServersArray;
+    }
+    if (!options.turn_disabled_remote && !options.config_remote.iceServers) {
+      options.config_remote.iceServers = iceServersArray;
+    }
+  } else if (typeof turnServers !== "undefined") {
+    if (!options.turn_disabled_local && turnServers.local) {
+      if (!options.config_local.hasOwnProperty("iceServers")) {
+        options.config_local.iceServers = turnServers.local.iceServers;
+      }
+    }
+    if (!options.turn_disabled_remote && turnServers.remote) {
+      if (!options.config_remote.hasOwnProperty("iceServers")) {
+        options.config_remote.iceServers = turnServers.remote.iceServers;
+      }
+    }
+  }
+
+  if (options.is_local) {
+    this.pcLocal = new PeerConnectionWrapper("pcLocal", options.config_local);
+  } else {
+    this.pcLocal = null;
+  }
+
+  if (options.is_remote) {
+    this.pcRemote = new PeerConnectionWrapper(
+      "pcRemote",
+      options.config_remote || options.config_local
+    );
+  } else {
+    this.pcRemote = null;
+  }
+
+  // Create command chain instance and assign default commands
+  this.chain = new CommandChain(this, options.commands);
+
+  this.testOptions = options;
+}
+
+/** TODO: consider removing this dependency on timeouts */
+function timerGuard(p, time, message) {
+  return Promise.race([
+    p,
+    wait(time).then(() => {
+      throw new Error("timeout after " + time / 1000 + "s: " + message);
+    }),
+  ]);
+}
+
+/**
+ * Closes the peer connection if it is active
+ */
+PeerConnectionTest.prototype.closePC = function () {
+  info("Closing peer connections");
+
+  var closeIt = pc => {
+    if (!pc || pc.signalingState === "closed") {
+      return Promise.resolve();
+    }
+
+    var promise = Promise.all([
+      Promise.all(
+        pc._pc
+          .getReceivers()
+          .filter(receiver => receiver.track.readyState == "live")
+          .map(receiver => {
+            info(
+              "Waiting for track " +
+                receiver.track.id +
+                " (" +
+                receiver.track.kind +
+                ") to end."
+            );
+            return haveEvent(receiver.track, "ended", wait(50000)).then(
+              event => {
+                is(
+                  event.target,
+                  receiver.track,
+                  "Event target should be the correct track"
+                );
+                info(pc + " ended fired for track " + receiver.track.id);
+              },
+              e =>
+                e
+                  ? Promise.reject(e)
+                  : ok(
+                      false,
+                      "ended never fired for track " + receiver.track.id
+                    )
+            );
+          })
+      ),
+    ]);
+    pc.close();
+    return promise;
+  };
+
+  return timerGuard(
+    Promise.all([closeIt(this.pcLocal), closeIt(this.pcRemote)]),
+    60000,
+    "failed to close peer connection"
+  );
+};
+
+/**
+ * Close the open data channels, followed by the underlying peer connection
+ */
+PeerConnectionTest.prototype.close = function () {
+  var allChannels = (this.pcLocal || this.pcRemote).dataChannels;
+  return timerGuard(
+    Promise.all(allChannels.map((channel, i) => this.closeDataChannels(i))),
+    120000,
+    "failed to close data channels"
+  ).then(() => this.closePC());
+};
+
+/**
+ * Close the specified data channels
+ *
+ * @param {Number} index
+ *        Index of the data channels to close on both sides
+ */
+PeerConnectionTest.prototype.closeDataChannels = function (index) {
+  info("closeDataChannels called with index: " + index);
+  var localChannel = null;
+  if (this.pcLocal) {
+    localChannel = this.pcLocal.dataChannels[index];
+  }
+  var remoteChannel = null;
+  if (this.pcRemote) {
+    remoteChannel = this.pcRemote.dataChannels[index];
+  }
+
+  // We need to setup all the close listeners before calling close
+  var setupClosePromise = channel => {
+    if (!channel) {
+      return Promise.resolve();
+    }
+    return new Promise(resolve => {
+      channel.onclose = () => {
+        is(
+          channel.readyState,
+          "closed",
+          name + " channel " + index + " closed"
+        );
+        resolve();
+      };
+    });
+  };
+
+  // make sure to setup close listeners before triggering any actions
+  var allClosed = Promise.all([
+    setupClosePromise(localChannel),
+    setupClosePromise(remoteChannel),
+  ]);
+  var complete = timerGuard(
+    allClosed,
+    120000,
+    "failed to close data channel pair"
+  );
+
+  // triggering close on one side should suffice
+  if (remoteChannel) {
+    remoteChannel.close();
+  } else if (localChannel) {
+    localChannel.close();
+  }
+
+  return complete;
+};
+
+/**
+ * Send data (message or blob) to the other peer
+ *
+ * @param {String|Blob} data
+ *        Data to send to the other peer. For Blobs the MIME type will be lost.
+ * @param {Object} [options={ }]
+ *        Options to specify the data channels to be used
+ * @param {DataChannelWrapper} [options.sourceChannel=pcLocal.dataChannels[length - 1]]
+ *        Data channel to use for sending the message
+ * @param {DataChannelWrapper} [options.targetChannel=pcRemote.dataChannels[length - 1]]
+ *        Data channel to use for receiving the message
+ */
+PeerConnectionTest.prototype.send = async function (data, options) {
+  options = options || {};
+  const source =
+    options.sourceChannel ||
+    this.pcLocal.dataChannels[this.pcLocal.dataChannels.length - 1];
+  const target =
+    options.targetChannel ||
+    this.pcRemote.dataChannels[this.pcRemote.dataChannels.length - 1];
+  source.bufferedAmountLowThreshold = options.bufferedAmountLowThreshold || 0;
+
+  const getSizeInBytes = d => {
+    if (d instanceof Blob) {
+      return d.size;
+    } else if (d instanceof ArrayBuffer) {
+      return d.byteLength;
+    } else if (d instanceof String || typeof d === "string") {
+      return new TextEncoder().encode(d).length;
+    } else {
+      ok(false);
+    }
+  };
+
+  const expectedSizeInBytes = getSizeInBytes(data);
+  const bufferedAmount = source.bufferedAmount;
+
+  source.send(data);
+  is(
+    source.bufferedAmount,
+    expectedSizeInBytes + bufferedAmount,
+    `Buffered amount should be ${expectedSizeInBytes}`
+  );
+
+  await new Promise(resolve => (source.onbufferedamountlow = resolve));
+
+  return new Promise(resolve => {
+    // Register event handler for the target channel
+    target.onmessage = e => {
+      is(
+        getSizeInBytes(e.data),
+        expectedSizeInBytes,
+        `Expected to receive the same number of bytes as we sent (${expectedSizeInBytes})`
+      );
+      resolve({ channel: target, data: e.data });
+    };
+  });
+};
+
+/**
+ * Create a data channel
+ *
+ * @param {Dict} options
+ *        Options for the data channel (see nsIPeerConnection)
+ */
+PeerConnectionTest.prototype.createDataChannel = function (options) {
+  var remotePromise;
+  if (!options.negotiated) {
+    this.pcRemote.expectDataChannel("pcRemote expected data channel");
+    remotePromise = this.pcRemote.nextDataChannel;
+  }
+
+  // Create the datachannel
+  var localChannel = this.pcLocal.createDataChannel(options);
+  var localPromise = localChannel.opened;
+
+  if (options.negotiated) {
+    remotePromise = localPromise.then(localChannel => {
+      // externally negotiated - we need to open from both ends
+      options.id = options.id || channel.id; // allow for no id on options
+      var remoteChannel = this.pcRemote.createDataChannel(options);
+      return remoteChannel.opened;
+    });
+  }
+
+  // pcRemote.observedNegotiationNeeded might be undefined if
+  // !options.negotiated, which means we just wait on pcLocal
+  return Promise.all([
+    this.pcLocal.observedNegotiationNeeded,
+    this.pcRemote.observedNegotiationNeeded,
+  ]).then(() => {
+    return Promise.all([localPromise, remotePromise]).then(result => {
+      return { local: result[0], remote: result[1] };
+    });
+  });
+};
+
+/**
+ * Creates an answer for the specified peer connection instance
+ * and automatically handles the failure case.
+ *
+ * @param {PeerConnectionWrapper} peer
+ *        The peer connection wrapper to run the command on
+ */
+PeerConnectionTest.prototype.createAnswer = function (peer) {
+  return peer.createAnswer().then(answer => {
+    // make a copy so this does not get updated with ICE candidates
+    this.originalAnswer = new RTCSessionDescription(
+      JSON.parse(JSON.stringify(answer))
+    );
+    return answer;
+  });
+};
+
+/**
+ * Creates an offer for the specified peer connection instance
+ * and automatically handles the failure case.
+ *
+ * @param {PeerConnectionWrapper} peer
+ *        The peer connection wrapper to run the command on
+ */
+PeerConnectionTest.prototype.createOffer = function (peer) {
+  return peer.createOffer().then(offer => {
+    // make a copy so this does not get updated with ICE candidates
+    this.originalOffer = new RTCSessionDescription(
+      JSON.parse(JSON.stringify(offer))
+    );
+    return offer;
+  });
+};
+
+/**
+ * Sets the local description for the specified peer connection instance
+ * and automatically handles the failure case.
+ *
+ * @param {PeerConnectionWrapper} peer
+          The peer connection wrapper to run the command on
+ * @param {RTCSessionDescriptionInit} desc
+ *        Session description for the local description request
+ */
+PeerConnectionTest.prototype.setLocalDescription = function (
+  peer,
+  desc,
+  stateExpected
+) {
+  var eventFired = new Promise(resolve => {
+    peer.onsignalingstatechange = e => {
+      info(peer + ": 'signalingstatechange' event received");
+      var state = e.target.signalingState;
+      if (stateExpected === state) {
+        peer.setLocalDescStableEventDate = new Date();
+        resolve();
+      } else {
+        ok(
+          false,
+          "This event has either already fired or there has been a " +
+            "mismatch between event received " +
+            state +
+            " and event expected " +
+            stateExpected
+        );
+      }
+    };
+  });
+
+  var stateChanged = peer.setLocalDescription(desc).then(() => {
+    peer.setLocalDescDate = new Date();
+  });
+
+  peer.endOfTrickleSdp = peer.endOfTrickleIce
+    .then(() => {
+      return peer._pc.localDescription;
+    })
+    .catch(e => ok(false, "Sending EOC message failed: " + e));
+
+  return Promise.all([eventFired, stateChanged]);
+};
+
+/**
+ * Sets the media constraints for both peer connection instances.
+ *
+ * @param {object} constraintsLocal
+ *        Media constrains for the local peer connection instance
+ * @param constraintsRemote
+ */
+PeerConnectionTest.prototype.setMediaConstraints = function (
+  constraintsLocal,
+  constraintsRemote
+) {
+  if (this.pcLocal) {
+    this.pcLocal.constraints = constraintsLocal;
+  }
+  if (this.pcRemote) {
+    this.pcRemote.constraints = constraintsRemote;
+  }
+};
+
+/**
+ * Sets the media options used on a createOffer call in the test.
+ *
+ * @param {object} options the media constraints to use on createOffer
+ */
+PeerConnectionTest.prototype.setOfferOptions = function (options) {
+  if (this.pcLocal) {
+    this.pcLocal.offerOptions = options;
+  }
+};
+
+/**
+ * Sets the remote description for the specified peer connection instance
+ * and automatically handles the failure case.
+ *
+ * @param {PeerConnectionWrapper} peer
+          The peer connection wrapper to run the command on
+ * @param {RTCSessionDescriptionInit} desc
+ *        Session description for the remote description request
+ */
+PeerConnectionTest.prototype.setRemoteDescription = function (
+  peer,
+  desc,
+  stateExpected
+) {
+  var eventFired = new Promise(resolve => {
+    peer.onsignalingstatechange = e => {
+      info(peer + ": 'signalingstatechange' event received");
+      var state = e.target.signalingState;
+      if (stateExpected === state) {
+        peer.setRemoteDescStableEventDate = new Date();
+        resolve();
+      } else {
+        ok(
+          false,
+          "This event has either already fired or there has been a " +
+            "mismatch between event received " +
+            state +
+            " and event expected " +
+            stateExpected
+        );
+      }
+    };
+  });
+
+  var stateChanged = peer.setRemoteDescription(desc).then(() => {
+    peer.setRemoteDescDate = new Date();
+    peer.checkMediaTracks();
+  });
+
+  return Promise.all([eventFired, stateChanged]);
+};
+
+/**
+ * Adds and removes steps to/from the execution chain based on the configured
+ * testOptions.
+ */
+PeerConnectionTest.prototype.updateChainSteps = function () {
+  if (this.testOptions.h264) {
+    this.chain.insertAfterEach("PC_LOCAL_CREATE_OFFER", [
+      PC_LOCAL_REMOVE_ALL_BUT_H264_FROM_OFFER,
+    ]);
+  }
+  if (!this.testOptions.bundle) {
+    this.chain.insertAfterEach("PC_LOCAL_CREATE_OFFER", [
+      PC_LOCAL_REMOVE_BUNDLE_FROM_OFFER,
+    ]);
+  }
+  if (!this.testOptions.rtcpmux) {
+    this.chain.insertAfterEach("PC_LOCAL_CREATE_OFFER", [
+      PC_LOCAL_REMOVE_RTCPMUX_FROM_OFFER,
+    ]);
+  }
+  if (!this.testOptions.ssrc) {
+    this.chain.insertAfterEach("PC_LOCAL_CREATE_OFFER", [
+      PC_LOCAL_REMOVE_SSRC_FROM_OFFER,
+    ]);
+    this.chain.insertAfterEach("PC_REMOTE_CREATE_ANSWER", [
+      PC_REMOTE_REMOVE_SSRC_FROM_ANSWER,
+    ]);
+  }
+  if (!this.testOptions.is_local) {
+    this.chain.filterOut(/^PC_LOCAL/);
+  }
+  if (!this.testOptions.is_remote) {
+    this.chain.filterOut(/^PC_REMOTE/);
+  }
+};
+
+/**
+ * Start running the tests as assigned to the command chain.
+ */
+PeerConnectionTest.prototype.run = async function () {
+  /* We have to modify the chain here to allow tests which modify the default
+   * test chain instantiating a PeerConnectionTest() */
+  this.updateChainSteps();
+  try {
+    await this.chain.execute();
+    await this.close();
+  } catch (e) {
+    const stack =
+      typeof e.stack === "string"
+        ? ` ${e.stack.split("\n").join(" ... ")}`
+        : "";
+    ok(false, `Error in test execution: ${e} (${stack})`);
+  }
+};
+
+/**
+ * Routes ice candidates from one PCW to the other PCW
+ */
+PeerConnectionTest.prototype.iceCandidateHandler = function (
+  caller,
+  candidate
+) {
+  info("Received: " + JSON.stringify(candidate) + " from " + caller);
+
+  var target = null;
+  if (caller.includes("pcLocal")) {
+    if (this.pcRemote) {
+      target = this.pcRemote;
+    }
+  } else if (caller.includes("pcRemote")) {
+    if (this.pcLocal) {
+      target = this.pcLocal;
+    }
+  } else {
+    ok(false, "received event from unknown caller: " + caller);
+    return;
+  }
+
+  if (target) {
+    target.storeOrAddIceCandidate(candidate);
+  } else {
+    info("sending ice candidate to signaling server");
+    send_message({ type: "ice_candidate", ice_candidate: candidate });
+  }
+};
+
+/**
+ * Installs a polling function for the socket.io client to read
+ * all messages from the chat room into a message queue.
+ */
+PeerConnectionTest.prototype.setupSignalingClient = function () {
+  this.signalingMessageQueue = [];
+  this.signalingCallbacks = {};
+  this.signalingLoopRun = true;
+
+  var queueMessage = message => {
+    info("Received signaling message: " + JSON.stringify(message));
+    var fired = false;
+    Object.keys(this.signalingCallbacks).forEach(name => {
+      if (name === message.type) {
+        info("Invoking callback for message type: " + name);
+        this.signalingCallbacks[name](message);
+        fired = true;
+      }
+    });
+    if (!fired) {
+      this.signalingMessageQueue.push(message);
+      info(
+        "signalingMessageQueue.length: " + this.signalingMessageQueue.length
+      );
+    }
+    if (this.signalingLoopRun) {
+      wait_for_message().then(queueMessage);
+    } else {
+      info("Exiting signaling message event loop");
+    }
+  };
+  wait_for_message().then(queueMessage);
+};
+
+/**
+ * Sets a flag to stop reading further messages from the chat room.
+ */
+PeerConnectionTest.prototype.signalingMessagesFinished = function () {
+  this.signalingLoopRun = false;
+};
+
+/**
+ * Register a callback function to deliver messages from the chat room
+ * directly instead of storing them in the message queue.
+ *
+ * @param {string} messageType
+ *        For which message types should the callback get invoked.
+ *
+ * @param {function} onMessage
+ *        The function which gets invoked if a message of the messageType
+ *        has been received from the chat room.
+ */
+PeerConnectionTest.prototype.registerSignalingCallback = function (
+  messageType,
+  onMessage
+) {
+  this.signalingCallbacks[messageType] = onMessage;
+};
+
+/**
+ * Searches the message queue for the first message of a given type
+ * and invokes the given callback function, or registers the callback
+ * function for future messages if the queue contains no such message.
+ *
+ * @param {string} messageType
+ *        The type of message to search and register for.
+ */
+PeerConnectionTest.prototype.getSignalingMessage = function (messageType) {
+  var i = this.signalingMessageQueue.findIndex(m => m.type === messageType);
+  if (i >= 0) {
+    info(
+      "invoking callback on message " +
+        i +
+        " from message queue, for message type:" +
+        messageType
+    );
+    return Promise.resolve(this.signalingMessageQueue.splice(i, 1)[0]);
+  }
+  return new Promise(resolve =>
+    this.registerSignalingCallback(messageType, resolve)
+  );
+};
+
+/**
+ * This class acts as a wrapper around a DataChannel instance.
+ *
+ * @param dataChannel
+ * @param peerConnectionWrapper
+ * @constructor
+ */
+function DataChannelWrapper(dataChannel, peerConnectionWrapper) {
+  this._channel = dataChannel;
+  this._pc = peerConnectionWrapper;
+
+  info("Creating " + this);
+
+  /**
+   * Setup appropriate callbacks
+   */
+  createOneShotEventWrapper(this, this._channel, "close");
+  createOneShotEventWrapper(this, this._channel, "error");
+  createOneShotEventWrapper(this, this._channel, "message");
+  createOneShotEventWrapper(this, this._channel, "bufferedamountlow");
+
+  this.opened = timerGuard(
+    new Promise(resolve => {
+      this._channel.onopen = () => {
+        this._channel.onopen = unexpectedEvent(this, "onopen");
+        is(this.readyState, "open", "data channel is 'open' after 'onopen'");
+        resolve(this);
+      };
+    }),
+    180000,
+    "channel didn't open in time"
+  );
+}
+
+DataChannelWrapper.prototype = {
+  /**
+   * Returns the binary type of the channel
+   *
+   * @returns {String} The binary type
+   */
+  get binaryType() {
+    return this._channel.binaryType;
+  },
+
+  /**
+   * Sets the binary type of the channel
+   *
+   * @param {String} type
+   *        The new binary type of the channel
+   */
+  set binaryType(type) {
+    this._channel.binaryType = type;
+  },
+
+  /**
+   * Returns the label of the underlying data channel
+   *
+   * @returns {String} The label
+   */
+  get label() {
+    return this._channel.label;
+  },
+
+  /**
+   * Returns the protocol of the underlying data channel
+   *
+   * @returns {String} The protocol
+   */
+  get protocol() {
+    return this._channel.protocol;
+  },
+
+  /**
+   * Returns the id of the underlying data channel
+   *
+   * @returns {number} The stream id
+   */
+  get id() {
+    return this._channel.id;
+  },
+
+  /**
+   * Returns the reliable state of the underlying data channel
+   *
+   * @returns {bool} The stream's reliable state
+   */
+  get reliable() {
+    return this._channel.reliable;
+  },
+
+  /**
+   * Returns the ordered attribute of the data channel
+   *
+   * @returns {bool} The ordered attribute
+   */
+  get ordered() {
+    return this._channel.ordered;
+  },
+
+  /**
+   * Returns the maxPacketLifeTime attribute of the data channel
+   *
+   * @returns {number} The maxPacketLifeTime attribute
+   */
+  get maxPacketLifeTime() {
+    return this._channel.maxPacketLifeTime;
+  },
+
+  /**
+   * Returns the maxRetransmits attribute of the data channel
+   *
+   * @returns {number} The maxRetransmits attribute
+   */
+  get maxRetransmits() {
+    return this._channel.maxRetransmits;
+  },
+
+  /**
+   * Returns the readyState bit of the data channel
+   *
+   * @returns {String} The state of the channel
+   */
+  get readyState() {
+    return this._channel.readyState;
+  },
+
+  get bufferedAmount() {
+    return this._channel.bufferedAmount;
+  },
+
+  /**
+   * Sets the bufferlowthreshold of the channel
+   *
+   * @param {integer} amoutn
+   *        The new threshold for the chanel
+   */
+  set bufferedAmountLowThreshold(amount) {
+    this._channel.bufferedAmountLowThreshold = amount;
+  },
+
+  /**
+   * Close the data channel
+   */
+  close() {
+    info(this + ": Closing channel");
+    this._channel.close();
+  },
+
+  /**
+   * Send data through the data channel
+   *
+   * @param {String|Object} data
+   *        Data which has to be sent through the data channel
+   */
+  send(data) {
+    info(this + ": Sending data '" + data + "'");
+    this._channel.send(data);
+  },
+
+  /**
+   * Returns the string representation of the class
+   *
+   * @returns {String} The string representation
+   */
+  toString() {
+    return (
+      "DataChannelWrapper (" + this._pc.label + "_" + this._channel.label + ")"
+    );
+  },
+};
+
+/**
+ * This class acts as a wrapper around a PeerConnection instance.
+ *
+ * @constructor
+ * @param {string} label
+ *        Description for the peer connection instance
+ * @param {object} configuration
+ *        Configuration for the peer connection instance
+ */
+function PeerConnectionWrapper(label, configuration) {
+  this.configuration = configuration;
+  if (configuration && configuration.label_suffix) {
+    label = label + "_" + configuration.label_suffix;
+  }
+  this.label = label;
+
+  this.constraints = [];
+  this.offerOptions = {};
+
+  this.dataChannels = [];
+
+  this._local_ice_candidates = [];
+  this._remote_ice_candidates = [];
+  this.localRequiresTrickleIce = false;
+  this.remoteRequiresTrickleIce = false;
+  this.localMediaElements = [];
+  this.remoteMediaElements = [];
+  this.audioElementsOnly = false;
+
+  this._sendStreams = [];
+
+  this.expectedLocalTrackInfo = [];
+  this.remoteStreamsByTrackId = new Map();
+
+  this.disableRtpCountChecking = false;
+
+  this.iceConnectedResolve;
+  this.iceConnectedReject;
+  this.iceConnected = new Promise((resolve, reject) => {
+    this.iceConnectedResolve = resolve;
+    this.iceConnectedReject = reject;
+  });
+  this.iceCheckingRestartExpected = false;
+  this.iceCheckingIceRollbackExpected = false;
+
+  info("Creating " + this);
+  this._pc = new RTCPeerConnection(this.configuration);
+
+  /**
+   * Setup callback handlers
+   */
+  // This allows test to register their own callbacks for ICE connection state changes
+  this.ice_connection_callbacks = {};
+
+  this._pc.oniceconnectionstatechange = e => {
+    isnot(
+      typeof this._pc.iceConnectionState,
+      "undefined",
+      "iceConnectionState should not be undefined"
+    );
+    var iceState = this._pc.iceConnectionState;
+    info(
+      this + ": oniceconnectionstatechange fired, new state is: " + iceState
+    );
+    Object.keys(this.ice_connection_callbacks).forEach(name => {
+      this.ice_connection_callbacks[name]();
+    });
+    if (iceState === "connected") {
+      this.iceConnectedResolve();
+    } else if (iceState === "failed") {
+      this.iceConnectedReject(new Error("ICE failed"));
+    }
+  };
+
+  this._pc.onicegatheringstatechange = e => {
+    isnot(
+      typeof this._pc.iceGatheringState,
+      "undefined",
+      "iceGetheringState should not be undefined"
+    );
+    var gatheringState = this._pc.iceGatheringState;
+    info(
+      this +
+        ": onicegatheringstatechange fired, new state is: " +
+        gatheringState
+    );
+  };
+
+  createOneShotEventWrapper(this, this._pc, "datachannel");
+  this._pc.addEventListener("datachannel", e => {
+    var wrapper = new DataChannelWrapper(e.channel, this);
+    this.dataChannels.push(wrapper);
+  });
+
+  createOneShotEventWrapper(this, this._pc, "signalingstatechange");
+  createOneShotEventWrapper(this, this._pc, "negotiationneeded");
+}
+
+PeerConnectionWrapper.prototype = {
+  /**
+   * Returns the senders
+   *
+   * @returns {sequence<RTCRtpSender>} the senders
+   */
+  getSenders() {
+    return this._pc.getSenders();
+  },
+
+  /**
+   * Returns the getters
+   *
+   * @returns {sequence<RTCRtpReceiver>} the receivers
+   */
+  getReceivers() {
+    return this._pc.getReceivers();
+  },
+
+  /**
+   * Returns the local description.
+   *
+   * @returns {object} The local description
+   */
+  get localDescription() {
+    return this._pc.localDescription;
+  },
+
+  /**
+   * Returns the remote description.
+   *
+   * @returns {object} The remote description
+   */
+  get remoteDescription() {
+    return this._pc.remoteDescription;
+  },
+
+  /**
+   * Returns the signaling state.
+   *
+   * @returns {object} The local description
+   */
+  get signalingState() {
+    return this._pc.signalingState;
+  },
+  /**
+   * Returns the ICE connection state.
+   *
+   * @returns {object} The local description
+   */
+  get iceConnectionState() {
+    return this._pc.iceConnectionState;
+  },
+
+  setIdentityProvider(provider, options) {
+    this._pc.setIdentityProvider(provider, options);
+  },
+
+  elementPrefix: direction => {
+    return [this.label, direction].join("_");
+  },
+
+  getMediaElementForTrack(track, direction) {
+    var prefix = this.elementPrefix(direction);
+    return getMediaElementForTrack(track, prefix);
+  },
+
+  createMediaElementForTrack(track, direction) {
+    var prefix = this.elementPrefix(direction);
+    return createMediaElementForTrack(track, prefix);
+  },
+
+  ensureMediaElement(track, direction) {
+    var prefix = this.elementPrefix(direction);
+    var element = this.getMediaElementForTrack(track, direction);
+    if (!element) {
+      element = this.createMediaElementForTrack(track, direction);
+      if (direction == "local") {
+        this.localMediaElements.push(element);
+      } else if (direction == "remote") {
+        this.remoteMediaElements.push(element);
+      }
+    }
+
+    // We do this regardless, because sometimes we end up with a new stream with
+    // an old id (ie; the rollback tests cause the same stream to be added
+    // twice)
+    element.srcObject = new MediaStream([track]);
+    element.play();
+  },
+
+  addSendStream(stream) {
+    // The PeerConnection will not necessarily know about this stream
+    // automatically, because replaceTrack is not told about any streams the
+    // new track might be associated with. Only content really knows.
+    this._sendStreams.push(stream);
+  },
+
+  getStreamForSendTrack(track) {
+    return this._sendStreams.find(str => str.getTrackById(track.id));
+  },
+
+  getStreamForRecvTrack(track) {
+    return this._pc.getRemoteStreams().find(s => !!s.getTrackById(track.id));
+  },
+
+  /**
+   * Attaches a local track to this RTCPeerConnection using
+   * RTCPeerConnection.addTrack().
+   *
+   * Also creates a media element playing a MediaStream containing all
+   * tracks that have been added to `stream` using `attachLocalTrack()`.
+   *
+   * @param {MediaStreamTrack} track
+   *        MediaStreamTrack to handle
+   * @param {MediaStream} stream
+   *        MediaStream to use as container for `track` on remote side
+   */
+  attachLocalTrack(track, stream) {
+    info("Got a local " + track.kind + " track");
+
+    this.expectNegotiationNeeded();
+    var sender = this._pc.addTrack(track, stream);
+    is(sender.track, track, "addTrack returns sender");
+    is(
+      this._pc.getSenders().pop(),
+      sender,
+      "Sender should be the last element in getSenders()"
+    );
+
+    ok(track.id, "track has id");
+    ok(track.kind, "track has kind");
+    ok(stream.id, "stream has id");
+    this.expectedLocalTrackInfo.push({ track, sender, streamId: stream.id });
+    this.addSendStream(stream);
+
+    // This will create one media element per track, which might not be how
+    // we set up things with the RTCPeerConnection. It's the only way
+    // we can ensure all sent tracks are flowing however.
+    this.ensureMediaElement(track, "local");
+
+    return this.observedNegotiationNeeded;
+  },
+
+  /**
+   * Callback when we get local media. Also an appropriate HTML media element
+   * will be created and added to the content node.
+   *
+   * @param {MediaStream} stream
+   *        Media stream to handle
+   */
+  attachLocalStream(stream, useAddTransceiver) {
+    info("Got local media stream: (" + stream.id + ")");
+
+    this.expectNegotiationNeeded();
+    if (useAddTransceiver) {
+      info("Using addTransceiver (on PC).");
+      stream.getTracks().forEach(track => {
+        var transceiver = this._pc.addTransceiver(track, { streams: [stream] });
+        is(transceiver.sender.track, track, "addTransceiver returns sender");
+      });
+    }
+    // In order to test both the addStream and addTrack APIs, we do half one
+    // way, half the other, at random.
+    else if (Math.random() < 0.5) {
+      info("Using addStream.");
+      this._pc.addStream(stream);
+      ok(
+        this._pc
+          .getSenders()
+          .find(sender => sender.track == stream.getTracks()[0]),
+        "addStream returns sender"
+      );
+    } else {
+      info("Using addTrack (on PC).");
+      stream.getTracks().forEach(track => {
+        var sender = this._pc.addTrack(track, stream);
+        is(sender.track, track, "addTrack returns sender");
+      });
+    }
+
+    this.addSendStream(stream);
+
+    stream.getTracks().forEach(track => {
+      ok(track.id, "track has id");
+      ok(track.kind, "track has kind");
+      const sender = this._pc.getSenders().find(s => s.track == track);
+      ok(sender, "track has a sender");
+      this.expectedLocalTrackInfo.push({ track, sender, streamId: stream.id });
+      this.ensureMediaElement(track, "local");
+    });
+
+    return this.observedNegotiationNeeded;
+  },
+
+  removeSender(index) {
+    var sender = this._pc.getSenders()[index];
+    this.expectedLocalTrackInfo = this.expectedLocalTrackInfo.filter(
+      i => i.sender != sender
+    );
+    this.expectNegotiationNeeded();
+    this._pc.removeTrack(sender);
+    return this.observedNegotiationNeeded;
+  },
+
+  senderReplaceTrack(sender, withTrack, stream) {
+    const info = this.expectedLocalTrackInfo.find(i => i.sender == sender);
+    if (!info) {
+      return; // replaceTrack on a null track, probably
+    }
+    info.track = withTrack;
+    this.addSendStream(stream);
+    this.ensureMediaElement(withTrack, "local");
+    return sender.replaceTrack(withTrack);
+  },
+
+  async getUserMedia(constraints) {
+    SpecialPowers.wrap(document).notifyUserGestureActivation();
+    var stream = await getUserMedia(constraints);
+    if (constraints.audio) {
+      stream.getAudioTracks().forEach(track => {
+        info(
+          this +
+            " gUM local stream " +
+            stream.id +
+            " with audio track " +
+            track.id
+        );
+      });
+    }
+    if (constraints.video) {
+      stream.getVideoTracks().forEach(track => {
+        info(
+          this +
+            " gUM local stream " +
+            stream.id +
+            " with video track " +
+            track.id
+        );
+      });
+    }
+    return stream;
+  },
+
+  /**
+   * Requests all the media streams as specified in the constrains property.
+   *
+   * @param {array} constraintsList
+   *        Array of constraints for GUM calls
+   */
+  getAllUserMedia(constraintsList) {
+    if (constraintsList.length === 0) {
+      info("Skipping GUM: no UserMedia requested");
+      return Promise.resolve();
+    }
+
+    info("Get " + constraintsList.length + " local streams");
+    return Promise.all(
+      constraintsList.map(constraints => this.getUserMedia(constraints))
+    );
+  },
+
+  async getAllUserMediaAndAddStreams(constraintsList) {
+    var streams = await this.getAllUserMedia(constraintsList);
+    if (!streams) {
+      return;
+    }
+    return Promise.all(streams.map(stream => this.attachLocalStream(stream)));
+  },
+
+  async getAllUserMediaAndAddTransceivers(constraintsList) {
+    var streams = await this.getAllUserMedia(constraintsList);
+    if (!streams) {
+      return;
+    }
+    return Promise.all(
+      streams.map(stream => this.attachLocalStream(stream, true))
+    );
+  },
+
+  /**
+   * Create a new data channel instance.  Also creates a promise called
+   * `this.nextDataChannel` that resolves when the next data channel arrives.
+   */
+  expectDataChannel(message) {
+    this.nextDataChannel = new Promise(resolve => {
+      this.ondatachannel = e => {
+        ok(e.channel, message);
+        is(
+          e.channel.readyState,
+          "open",
+          "data channel in 'open' after 'ondatachannel'"
+        );
+        resolve(e.channel);
+      };
+    });
+  },
+
+  /**
+   * Create a new data channel instance
+   *
+   * @param {Object} options
+   *        Options which get forwarded to nsIPeerConnection.createDataChannel
+   * @returns {DataChannelWrapper} The created data channel
+   */
+  createDataChannel(options) {
+    var label = "channel_" + this.dataChannels.length;
+    info(this + ": Create data channel '" + label);
+
+    if (!this.dataChannels.length) {
+      this.expectNegotiationNeeded();
+    }
+    var channel = this._pc.createDataChannel(label, options);
+    is(channel.readyState, "connecting", "initial readyState is 'connecting'");
+    var wrapper = new DataChannelWrapper(channel, this);
+    this.dataChannels.push(wrapper);
+    return wrapper;
+  },
+
+  /**
+   * Creates an offer and automatically handles the failure case.
+   */
+  createOffer() {
+    return this._pc.createOffer(this.offerOptions).then(offer => {
+      info("Got offer: " + JSON.stringify(offer));
+      // note: this might get updated through ICE gathering
+      this._latest_offer = offer;
+      return offer;
+    });
+  },
+
+  /**
+   * Creates an answer and automatically handles the failure case.
+   */
+  createAnswer() {
+    return this._pc.createAnswer().then(answer => {
+      info(this + ": Got answer: " + JSON.stringify(answer));
+      this._last_answer = answer;
+      return answer;
+    });
+  },
+
+  /**
+   * Sets the local description and automatically handles the failure case.
+   *
+   * @param {object} desc
+   *        RTCSessionDescriptionInit for the local description request
+   */
+  setLocalDescription(desc) {
+    this.observedNegotiationNeeded = undefined;
+    return this._pc.setLocalDescription(desc).then(() => {
+      info(this + ": Successfully set the local description");
+    });
+  },
+
+  /**
+   * Tries to set the local description and expect failure. Automatically
+   * causes the test case to fail if the call succeeds.
+   *
+   * @param {object} desc
+   *        RTCSessionDescriptionInit for the local description request
+   * @returns {Promise}
+   *        A promise that resolves to the expected error
+   */
+  setLocalDescriptionAndFail(desc) {
+    return this._pc
+      .setLocalDescription(desc)
+      .then(
+        generateErrorCallback("setLocalDescription should have failed."),
+        err => {
+          info(this + ": As expected, failed to set the local description");
+          return err;
+        }
+      );
+  },
+
+  /**
+   * Sets the remote description and automatically handles the failure case.
+   *
+   * @param {object} desc
+   *        RTCSessionDescriptionInit for the remote description request
+   */
+  setRemoteDescription(desc) {
+    this.observedNegotiationNeeded = undefined;
+    // This has to be done before calling sRD, otherwise a candidate in flight
+    // could end up in the PC's operations queue before sRD resolves.
+    if (desc.type == "rollback") {
+      this.holdIceCandidates = new Promise(
+        r => (this.releaseIceCandidates = r)
+      );
+    }
+    return this._pc.setRemoteDescription(desc).then(() => {
+      info(this + ": Successfully set remote description");
+      if (desc.type != "rollback") {
+        this.releaseIceCandidates();
+      }
+    });
+  },
+
+  /**
+   * Tries to set the remote description and expect failure. Automatically
+   * causes the test case to fail if the call succeeds.
+   *
+   * @param {object} desc
+   *        RTCSessionDescriptionInit for the remote description request
+   * @returns {Promise}
+   *        a promise that resolve to the returned error
+   */
+  setRemoteDescriptionAndFail(desc) {
+    return this._pc
+      .setRemoteDescription(desc)
+      .then(
+        generateErrorCallback("setRemoteDescription should have failed."),
+        err => {
+          info(this + ": As expected, failed to set the remote description");
+          return err;
+        }
+      );
+  },
+
+  /**
+   * Registers a callback for the signaling state change and
+   * appends the new state to an array for logging it later.
+   */
+  logSignalingState() {
+    this.signalingStateLog = [this._pc.signalingState];
+    this._pc.addEventListener("signalingstatechange", e => {
+      var newstate = this._pc.signalingState;
+      var oldstate = this.signalingStateLog[this.signalingStateLog.length - 1];
+      if (Object.keys(signalingStateTransitions).includes(oldstate)) {
+        ok(
+          signalingStateTransitions[oldstate].includes(newstate),
+          this +
+            ": legal signaling state transition from " +
+            oldstate +
+            " to " +
+            newstate
+        );
+      } else {
+        ok(
+          false,
+          this +
+            ": old signaling state " +
+            oldstate +
+            " missing in signaling transition array"
+        );
+      }
+      this.signalingStateLog.push(newstate);
+    });
+  },
+
+  isTrackOnPC(track) {
+    return !!this.getStreamForRecvTrack(track);
+  },
+
+  allExpectedTracksAreObserved(expected, observed) {
+    return Object.keys(expected).every(trackId => observed[trackId]);
+  },
+
+  setupStreamEventHandlers(stream) {
+    const myTrackIds = new Set(stream.getTracks().map(t => t.id));
+
+    stream.addEventListener("addtrack", ({ track }) => {
+      ok(
+        !myTrackIds.has(track.id),
+        "Duplicate addtrack callback: " +
+          `stream id=${stream.id} track id=${track.id}`
+      );
+      myTrackIds.add(track.id);
+      // addtrack events happen before track events, so the track callback hasn't
+      // heard about this yet.
+      let streams = this.remoteStreamsByTrackId.get(track.id);
+      ok(
+        !streams || !streams.has(stream.id),
+        `In addtrack for stream id=${stream.id}` +
+          `there should not have been a track event for track id=${track.id} ` +
+          " containing this stream yet."
+      );
+      ok(
+        stream.getTracks().includes(track),
+        "In addtrack, stream id=" +
+          `${stream.id} should already contain track id=${track.id}`
+      );
+    });
+
+    stream.addEventListener("removetrack", ({ track }) => {
+      ok(
+        myTrackIds.has(track.id),
+        "Duplicate removetrack callback: " +
+          `stream id=${stream.id} track id=${track.id}`
+      );
+      myTrackIds.delete(track.id);
+      // Also remove the association from remoteStreamsByTrackId
+      const streams = this.remoteStreamsByTrackId.get(track.id);
+      ok(
+        streams,
+        `In removetrack for stream id=${stream.id}, track id=` +
+          `${track.id} should have had a track callback for the stream.`
+      );
+      streams.delete(stream.id);
+      ok(
+        !stream.getTracks().includes(track),
+        "In removetrack, stream id=" +
+          `${stream.id} should not contain track id=${track.id}`
+      );
+    });
+  },
+
+  setupTrackEventHandler() {
+    this._pc.addEventListener("track", ({ track, streams }) => {
+      info(`${this}: 'ontrack' event fired for ${track.id}`);
+      ok(this.isTrackOnPC(track), `Found track ${track.id}`);
+
+      let gratuitousEvent = true;
+      let streamsContainingTrack = this.remoteStreamsByTrackId.get(track.id);
+      if (!streamsContainingTrack) {
+        gratuitousEvent = false; // Told us about a new track
+        this.remoteStreamsByTrackId.set(track.id, new Set());
+        streamsContainingTrack = this.remoteStreamsByTrackId.get(track.id);
+      }
+
+      for (const stream of streams) {
+        ok(
+          stream.getTracks().includes(track),
+          `In track event, track id=${track.id}` +
+            ` should already be in stream id=${stream.id}`
+        );
+
+        if (!streamsContainingTrack.has(stream.id)) {
+          gratuitousEvent = false; // Told us about a new stream
+          streamsContainingTrack.add(stream.id);
+          this.setupStreamEventHandlers(stream);
+        }
+      }
+
+      ok(!gratuitousEvent, "track event told us something new");
+
+      // So far, we've verified consistency between the current state of the
+      // streams, addtrack/removetrack events on the streams, and track events
+      // on the peerconnection. We have also verified that we have not gotten
+      // any gratuitous events. We have not done anything to verify that the
+      // current state of affairs matches what we were expecting it to.
+
+      this.ensureMediaElement(track, "remote");
+    });
+  },
+
+  /**
+   * Either adds a given ICE candidate right away or stores it to be added
+   * later, depending on the state of the PeerConnection.
+   *
+   * @param {object} candidate
+   *        The RTCIceCandidate to be added or stored
+   */
+  storeOrAddIceCandidate(candidate) {
+    this._remote_ice_candidates.push(candidate);
+    if (this.signalingState === "closed") {
+      info("Received ICE candidate for closed PeerConnection - discarding");
+      return;
+    }
+    this.holdIceCandidates
+      .then(() => {
+        info(this + ": adding ICE candidate " + JSON.stringify(candidate));
+        return this._pc.addIceCandidate(candidate);
+      })
+      .then(() => ok(true, this + " successfully added an ICE candidate"))
+      .catch(e =>
+        // The onicecandidate callback runs independent of the test steps
+        // and therefore errors thrown from in there don't get caught by the
+        // race of the Promises around our test steps.
+        // Note: as long as we are queuing ICE candidates until the success
+        //       of sRD() this should never ever happen.
+        ok(false, this + " adding ICE candidate failed with: " + e.message)
+      );
+  },
+
+  /**
+   * Registers a callback for the ICE connection state change and
+   * appends the new state to an array for logging it later.
+   */
+  logIceConnectionState() {
+    this.iceConnectionLog = [this._pc.iceConnectionState];
+    this.ice_connection_callbacks.logIceStatus = () => {
+      var newstate = this._pc.iceConnectionState;
+      var oldstate = this.iceConnectionLog[this.iceConnectionLog.length - 1];
+      if (Object.keys(iceStateTransitions).includes(oldstate)) {
+        if (this.iceCheckingRestartExpected) {
+          is(
+            newstate,
+            "checking",
+            "iceconnectionstate event '" +
+              newstate +
+              "' matches expected state 'checking'"
+          );
+          this.iceCheckingRestartExpected = false;
+        } else if (this.iceCheckingIceRollbackExpected) {
+          is(
+            newstate,
+            "connected",
+            "iceconnectionstate event '" +
+              newstate +
+              "' matches expected state 'connected'"
+          );
+          this.iceCheckingIceRollbackExpected = false;
+        } else {
+          ok(
+            iceStateTransitions[oldstate].includes(newstate),
+            this +
+              ": legal ICE state transition from " +
+              oldstate +
+              " to " +
+              newstate
+          );
+        }
+      } else {
+        ok(
+          false,
+          this +
+            ": old ICE state " +
+            oldstate +
+            " missing in ICE transition array"
+        );
+      }
+      this.iceConnectionLog.push(newstate);
+    };
+  },
+
+  /**
+   * Resets the ICE connected Promise and allows ICE connection state monitoring
+   * to go backwards to 'checking'.
+   */
+  expectIceChecking() {
+    this.iceCheckingRestartExpected = true;
+    this.iceConnected = new Promise((resolve, reject) => {
+      this.iceConnectedResolve = resolve;
+      this.iceConnectedReject = reject;
+    });
+  },
+
+  /**
+   * Waits for ICE to either connect or fail.
+   *
+   * @returns {Promise}
+   *          resolves when connected, rejects on failure
+   */
+  waitForIceConnected() {
+    return this.iceConnected;
+  },
+
+  /**
+   * Setup a onicecandidate handler
+   *
+   * @param {object} test
+   *        A PeerConnectionTest object to which the ice candidates gets
+   *        forwarded.
+   */
+  setupIceCandidateHandler(test, candidateHandler) {
+    candidateHandler = candidateHandler || test.iceCandidateHandler.bind(test);
+
+    var resolveEndOfTrickle;
+    this.endOfTrickleIce = new Promise(r => (resolveEndOfTrickle = r));
+    this.holdIceCandidates = new Promise(r => (this.releaseIceCandidates = r));
+
+    this._pc.onicecandidate = anEvent => {
+      if (!anEvent.candidate) {
+        this._pc.onicecandidate = () =>
+          ok(
+            false,
+            this.label + " received ICE candidate after end of trickle"
+          );
+        info(this.label + ": received end of trickle ICE event");
+        ok(
+          this._pc.iceGatheringState === "complete",
+          "ICE gathering state has reached complete"
+        );
+        resolveEndOfTrickle(this.label);
+        return;
+      }
+
+      info(
+        this.label + ": iceCandidate = " + JSON.stringify(anEvent.candidate)
+      );
+      ok(anEvent.candidate.sdpMid.length, "SDP mid not empty");
+      ok(
+        anEvent.candidate.usernameFragment.length,
+        "usernameFragment not empty"
+      );
+
+      ok(
+        typeof anEvent.candidate.sdpMLineIndex === "number",
+        "SDP MLine Index needs to exist"
+      );
+      this._local_ice_candidates.push(anEvent.candidate);
+      candidateHandler(this.label, anEvent.candidate);
+    };
+  },
+
+  checkLocalMediaTracks() {
+    info(
+      `${this}: Checking local tracks ${JSON.stringify(
+        this.expectedLocalTrackInfo
+      )}`
+    );
+    const sendersWithTrack = this._pc.getSenders().filter(({ track }) => track);
+    is(
+      sendersWithTrack.length,
+      this.expectedLocalTrackInfo.length,
+      "The number of senders with a track should be equal to the number of " +
+        "expected local tracks."
+    );
+
+    // expectedLocalTrackInfo is in the same order that the tracks were added, and
+    // so should the output of getSenders.
+    this.expectedLocalTrackInfo.forEach((info, i) => {
+      const sender = sendersWithTrack[i];
+      is(sender, info.sender, `Sender ${i} should match`);
+      is(sender.track, info.track, `Track ${i} should match`);
+    });
+  },
+
+  /**
+   * Checks that we are getting the media tracks we expect.
+   */
+  checkMediaTracks() {
+    this.checkLocalMediaTracks();
+  },
+
+  checkLocalMsids() {
+    const sdp = this.localDescription.sdp;
+    const msections = sdputils.getMSections(sdp);
+    const expectedStreamIdCounts = new Map();
+    for (const { track, sender, streamId } of this.expectedLocalTrackInfo) {
+      const transceiver = this._pc
+        .getTransceivers()
+        .find(t => t.sender == sender);
+      ok(transceiver, "There should be a transceiver for each sender");
+      if (transceiver.mid) {
+        const midFinder = new RegExp(`^a=mid:${transceiver.mid}$`, "m");
+        const msection = msections.find(m => m.match(midFinder));
+        ok(
+          msection,
+          `There should be a media section for mid = ${transceiver.mid}`
+        );
+        ok(
+          msection.startsWith(`m=${track.kind}`),
+          `Media section should be of type ${track.kind}`
+        );
+        const msidFinder = new RegExp(`^a=msid:${streamId} \\S+$`, "m");
+        ok(
+          msection.match(msidFinder),
+          `Should find a=msid:${streamId} in media section` +
+            " (with any track id for now)"
+        );
+        const count = expectedStreamIdCounts.get(streamId) || 0;
+        expectedStreamIdCounts.set(streamId, count + 1);
+      }
+    }
+
+    // Check for any unexpected msids.
+    const allMsids = sdp.match(new RegExp("^a=msid:\\S+", "mg"));
+    if (!allMsids) {
+      return;
+    }
+    const allStreamIds = allMsids.map(msidAttr =>
+      msidAttr.replace("a=msid:", "")
+    );
+    allStreamIds.forEach(id => {
+      const count = expectedStreamIdCounts.get(id);
+      ok(count, `Unexpected stream id ${id} found in local description.`);
+      if (count) {
+        expectedStreamIdCounts.set(id, count - 1);
+      }
+    });
+  },
+
+  /**
+   * Check that media flow is present for the given media element by checking
+   * that it reaches ready state HAVE_ENOUGH_DATA and progresses time further
+   * than the start of the check.
+   *
+   * This ensures, that the stream being played is producing
+   * data and, in case it contains a video track, that at least one video frame
+   * has been displayed.
+   *
+   * @param {HTMLMediaElement} track
+   *        The media element to check
+   * @returns {Promise}
+   *        A promise that resolves when media data is flowing.
+   */
+  waitForMediaElementFlow(element) {
+    info("Checking data flow for element: " + element.id);
+    is(
+      element.ended,
+      !element.srcObject.active,
+      "Element ended should be the inverse of the MediaStream's active state"
+    );
+    if (element.ended) {
+      is(
+        element.readyState,
+        element.HAVE_CURRENT_DATA,
+        "Element " + element.id + " is ended and should have had data"
+      );
+      return Promise.resolve();
+    }
+
+    const haveEnoughData = (
+      element.readyState == element.HAVE_ENOUGH_DATA
+        ? Promise.resolve()
+        : haveEvent(
+            element,
+            "canplay",
+            wait(60000, new Error("Timeout for element " + element.id))
+          )
+    ).then(_ => info("Element " + element.id + " has enough data."));
+
+    const startTime = element.currentTime;
+    const timeProgressed = timeout(
+      listenUntil(element, "timeupdate", _ => element.currentTime > startTime),
+      60000,
+      "Element " + element.id + " should progress currentTime"
+    ).then();
+
+    return Promise.all([haveEnoughData, timeProgressed]);
+  },
+
+  /**
+   * Wait for RTP packet flow for the given MediaStreamTrack.
+   *
+   * @param {object} track
+   *        A MediaStreamTrack to wait for data flow on.
+   * @returns {Promise}
+   *        Returns a promise which yields a StatsReport object with RTP stats.
+   */
+  async _waitForRtpFlow(target, rtpType) {
+    const { track } = target;
+    info(`_waitForRtpFlow(${track.id}, ${rtpType})`);
+    const packets = `packets${rtpType == "outbound-rtp" ? "Sent" : "Received"}`;
+
+    const retryInterval = 500; // Time between stats checks
+    const timeout = 30000; // Timeout in ms
+    const retries = timeout / retryInterval;
+
+    for (let i = 0; i < retries; i++) {
+      info(`Checking ${rtpType} for ${track.kind} track ${track.id} try ${i}`);
+      for (const rtp of (await target.getStats()).values()) {
+        if (rtp.type != rtpType) {
+          continue;
+        }
+        if (rtp.kind != track.kind) {
+          continue;
+        }
+
+        const numPackets = rtp[packets];
+        info(`Track ${track.id} has ${numPackets} ${packets}.`);
+        if (!numPackets) {
+          continue;
+        }
+
+        ok(true, `RTP flowing for ${track.kind} track ${track.id}`);
+        return;
+      }
+      await wait(retryInterval);
+    }
+    throw new Error(
+      `Checking stats for track ${track.id} timed out after ${timeout} ms`
+    );
+  },
+
+  /**
+   * Wait for inbound RTP packet flow for the given MediaStreamTrack.
+   *
+   * @param {object} receiver
+   *        An RTCRtpReceiver to wait for data flow on.
+   * @returns {Promise}
+   *        Returns a promise that resolves once data is flowing.
+   */
+  async waitForInboundRtpFlow(receiver) {
+    return this._waitForRtpFlow(receiver, "inbound-rtp");
+  },
+
+  /**
+   * Wait for outbound RTP packet flow for the given MediaStreamTrack.
+   *
+   * @param {object} sender
+   *        An RTCRtpSender to wait for data flow on.
+   * @returns {Promise}
+   *        Returns a promise that resolves once data is flowing.
+   */
+  async waitForOutboundRtpFlow(sender) {
+    return this._waitForRtpFlow(sender, "outbound-rtp");
+  },
+
+  getExpectedActiveReceivers() {
+    return this._pc
+      .getTransceivers()
+      .filter(
+        t =>
+          !t.stopped &&
+          t.currentDirection &&
+          t.currentDirection != "inactive" &&
+          t.currentDirection != "sendonly"
+      )
+      .filter(({ receiver }) => receiver.track)
+      .map(({ mid, currentDirection, receiver }) => {
+        info(
+          `Found transceiver that should be receiving RTP: mid=${mid}` +
+            ` currentDirection=${currentDirection}` +
+            ` kind=${receiver.track.kind} track-id=${receiver.track.id}`
+        );
+        return receiver;
+      });
+  },
+
+  getExpectedSenders() {
+    return this._pc.getSenders().filter(({ track }) => track);
+  },
+
+  /**
+   * Wait for presence of video flow on all media elements and rtp flow on
+   * all sending and receiving track involved in this test.
+   *
+   * @returns {Promise}
+   *        A promise that resolves when media flows for all elements and tracks
+   */
+  waitForMediaFlow() {
+    const receivers = this.getExpectedActiveReceivers();
+    return Promise.all([
+      ...this.localMediaElements.map(el => this.waitForMediaElementFlow(el)),
+      ...this.remoteMediaElements
+        .filter(({ srcObject }) =>
+          receivers.some(({ track }) =>
+            srcObject.getTracks().some(t => t == track)
+          )
+        )
+        .map(el => this.waitForMediaElementFlow(el)),
+      ...receivers.map(receiver => this.waitForInboundRtpFlow(receiver)),
+      ...this.getExpectedSenders().map(sender =>
+        this.waitForOutboundRtpFlow(sender)
+      ),
+    ]);
+  },
+
+  /**
+   * Check that correct audio (typically a flat tone) is flowing to this
+   * PeerConnection for each transceiver that should be receiving. Uses
+   * WebAudio AnalyserNodes to compare input and output audio data in the
+   * frequency domain.
+   *
+   * @param {object} from
+   *        A PeerConnectionWrapper whose audio RTPSender we use as source for
+   *        the audio flow check.
+   * @returns {Promise}
+   *        A promise that resolves when we're receiving the tone/s from |from|.
+   */
+  async checkReceivingToneFrom(
+    audiocontext,
+    from,
+    cancel = wait(60000, new Error("Tone not detected"))
+  ) {
+    let localTransceivers = this._pc
+      .getTransceivers()
+      .filter(t => t.mid)
+      .filter(t => t.receiver.track.kind == "audio")
+      .sort((t1, t2) => t1.mid < t2.mid);
+    let remoteTransceivers = from._pc
+      .getTransceivers()
+      .filter(t => t.mid)
+      .filter(t => t.receiver.track.kind == "audio")
+      .sort((t1, t2) => t1.mid < t2.mid);
+
+    is(
+      localTransceivers.length,
+      remoteTransceivers.length,
+      "Same number of associated audio transceivers on remote and local."
+    );
+
+    for (let i = 0; i < localTransceivers.length; i++) {
+      is(
+        localTransceivers[i].mid,
+        remoteTransceivers[i].mid,
+        "Transceivers at index " + i + " have the same mid."
+      );
+
+      if (!remoteTransceivers[i].sender.track) {
+        continue;
+      }
+
+      if (
+        remoteTransceivers[i].currentDirection == "recvonly" ||
+        remoteTransceivers[i].currentDirection == "inactive"
+      ) {
+        continue;
+      }
+
+      let sendTrack = remoteTransceivers[i].sender.track;
+      let inputElem = from.getMediaElementForTrack(sendTrack, "local");
+      ok(
+        inputElem,
+        "Remote wrapper should have a media element for track id " +
+          sendTrack.id
+      );
+      let inputAudioStream = from.getStreamForSendTrack(sendTrack);
+      ok(
+        inputAudioStream,
+        "Remote wrapper should have a stream for track id " + sendTrack.id
+      );
+      let inputAnalyser = new AudioStreamAnalyser(
+        audiocontext,
+        inputAudioStream
+      );
+
+      let recvTrack = localTransceivers[i].receiver.track;
+      let outputAudioStream = this.getStreamForRecvTrack(recvTrack);
+      ok(
+        outputAudioStream,
+        "Local wrapper should have a stream for track id " + recvTrack.id
+      );
+      let outputAnalyser = new AudioStreamAnalyser(
+        audiocontext,
+        outputAudioStream
+      );
+
+      let error = null;
+      cancel.then(e => (error = e));
+
+      let indexOfMax = data =>
+        data.reduce((max, val, i) => (val >= data[max] ? i : max), 0);
+
+      await outputAnalyser.waitForAnalysisSuccess(() => {
+        if (error) {
+          throw error;
+        }
+
+        let inputData = inputAnalyser.getByteFrequencyData();
+        let outputData = outputAnalyser.getByteFrequencyData();
+
+        let inputMax = indexOfMax(inputData);
+        let outputMax = indexOfMax(outputData);
+        info(
+          `Comparing maxima; input[${inputMax}] = ${inputData[inputMax]},` +
+            ` output[${outputMax}] = ${outputData[outputMax]}`
+        );
+        if (!inputData[inputMax] || !outputData[outputMax]) {
+          return false;
+        }
+
+        // When the input and output maxima are within reasonable distance (2% of
+        // total length, which means ~10 for length 512) from each other, we can
+        // be sure that the input tone has made it through the peer connection.
+        info(`input data length: ${inputData.length}`);
+        return Math.abs(inputMax - outputMax) < inputData.length * 0.02;
+      });
+    }
+  },
+
+  /**
+   * Check that stats are present by checking for known stats.
+   */
+  async getStats(selector) {
+    const stats = await this._pc.getStats(selector);
+    const dict = {};
+    for (const [k, v] of stats.entries()) {
+      dict[k] = v;
+    }
+    info(`${this}: Got stats: ${JSON.stringify(dict)}`);
+    return stats;
+  },
+
+  /**
+   * Checks that we are getting the media streams we expect.
+   *
+   * @param {object} stats
+   *        The stats to check from this PeerConnectionWrapper
+   */
+  checkStats(stats) {
+    const isRemote = ({ type }) =>
+      ["remote-outbound-rtp", "remote-inbound-rtp"].includes(type);
+    var counters = {};
+    for (let [key, res] of stats) {
+      info("Checking stats for " + key + " : " + res);
+      // validate stats
+      ok(res.id == key, "Coherent stats id");
+      const now = performance.timeOrigin + performance.now();
+      const minimum = performance.timeOrigin;
+      const type = isRemote(res) ? "rtcp" : "rtp";
+      ok(
+        res.timestamp >= minimum,
+        `Valid ${type} timestamp ${res.timestamp} >= ${minimum} (
+            ${res.timestamp - minimum} ms)`
+      );
+      ok(
+        res.timestamp <= now,
+        `Valid ${type} timestamp ${res.timestamp} <= ${now} (
+            ${res.timestamp - now} ms)`
+      );
+      if (isRemote(res)) {
+        continue;
+      }
+      counters[res.type] = (counters[res.type] || 0) + 1;
+
+      switch (res.type) {
+        case "inbound-rtp":
+        case "outbound-rtp":
+          {
+            // Inbound tracks won't have an ssrc if RTP is not flowing.
+            // (eg; negotiated inactive)
+            ok(
+              res.ssrc || res.type == "inbound-rtp",
+              "Outbound RTP stats has an ssrc."
+            );
+
+            if (res.ssrc) {
+              // ssrc is a 32 bit number returned as an unsigned long
+              ok(!/[^0-9]/.test(`${res.ssrc}`), "SSRC is numeric");
+              ok(parseInt(res.ssrc) < Math.pow(2, 32), "SSRC is within limits");
+            }
+
+            if (res.type == "outbound-rtp") {
+              ok(res.packetsSent !== undefined, "Rtp packetsSent");
+              // We assume minimum payload to be 1 byte (guess from RFC 3550)
+              ok(res.bytesSent >= res.packetsSent, "Rtp bytesSent");
+            } else {
+              ok(res.packetsReceived !== undefined, "Rtp packetsReceived");
+              ok(res.bytesReceived >= res.packetsReceived, "Rtp bytesReceived");
+            }
+            if (res.remoteId) {
+              var rem = stats.get(res.remoteId);
+              ok(isRemote(rem), "Remote is rtcp");
+              ok(rem.localId == res.id, "Remote backlink match");
+              if (res.type == "outbound-rtp") {
+                ok(rem.type == "remote-inbound-rtp", "Rtcp is inbound");
+                if (rem.packetsLost) {
+                  ok(
+                    rem.packetsLost >= 0,
+                    "Rtcp packetsLost " + rem.packetsLost + " >= 0"
+                  );
+                  ok(
+                    rem.packetsLost < 1000,
+                    "Rtcp packetsLost " + rem.packetsLost + " < 1000"
+                  );
+                }
+                if (!this.disableRtpCountChecking) {
+                  // no guarantee which one is newer!
+                  // Note: this must change when we add a timestamp field to remote RTCP reports
+                  // and make rem.timestamp be the reception time
+                  if (res.timestamp < rem.timestamp) {
+                    info(
+                      "REVERSED timestamps: rec:" +
+                        rem.packetsReceived +
+                        " time:" +
+                        rem.timestamp +
+                        " sent:" +
+                        res.packetsSent +
+                        " time:" +
+                        res.timestamp
+                    );
+                  }
+                }
+                if (rem.jitter) {
+                  ok(rem.jitter >= 0, "Rtcp jitter " + rem.jitter + " >= 0");
+                  ok(rem.jitter < 5, "Rtcp jitter " + rem.jitter + " < 5 sec");
+                }
+                if (rem.roundTripTime) {
+                  ok(
+                    rem.roundTripTime >= 0,
+                    "Rtcp rtt " + rem.roundTripTime + " >= 0"
+                  );
+                  ok(
+                    rem.roundTripTime < 60,
+                    "Rtcp rtt " + rem.roundTripTime + " < 1 min"
+                  );
+                }
+              } else {
+                ok(rem.type == "remote-outbound-rtp", "Rtcp is outbound");
+                ok(rem.packetsSent !== undefined, "Rtcp packetsSent");
+                ok(rem.bytesSent !== undefined, "Rtcp bytesSent");
+              }
+              ok(rem.ssrc == res.ssrc, "Remote ssrc match");
+            } else {
+              info("No rtcp info received yet");
+            }
+          }
+          break;
+      }
+    }
+
+    var nin = this._pc.getTransceivers().filter(t => {
+      return (
+        !t.stopped &&
+        t.currentDirection != "inactive" &&
+        t.currentDirection != "sendonly"
+      );
+    }).length;
+    const nout = Object.keys(this.expectedLocalTrackInfo).length;
+    var ndata = this.dataChannels.length;
+
+    // TODO(Bug 957145): Restore stronger inbound-rtp test once Bug 948249 is fixed
+    //is((counters["inbound-rtp"] || 0), nin, "Have " + nin + " inbound-rtp stat(s)");
+    ok(
+      (counters["inbound-rtp"] || 0) >= nin,
+      "Have at least " + nin + " inbound-rtp stat(s) *"
+    );
+
+    is(
+      counters["outbound-rtp"] || 0,
+      nout,
+      "Have " + nout + " outbound-rtp stat(s)"
+    );
+
+    var numLocalCandidates = counters["local-candidate"] || 0;
+    var numRemoteCandidates = counters["remote-candidate"] || 0;
+    // If there are no tracks, there will be no stats either.
+    if (nin + nout + ndata > 0) {
+      ok(numLocalCandidates, "Have local-candidate stat(s)");
+      ok(numRemoteCandidates, "Have remote-candidate stat(s)");
+    } else {
+      is(numLocalCandidates, 0, "Have no local-candidate stats");
+      is(numRemoteCandidates, 0, "Have no remote-candidate stats");
+    }
+  },
+
+  /**
+   * Compares the Ice server configured for this PeerConnectionWrapper
+   * with the ICE candidates received in the RTCP stats.
+   *
+   * @param {object} stats
+   *        The stats to be verified for relayed vs. direct connection.
+   */
+  checkStatsIceConnectionType(stats, expectedLocalCandidateType) {
+    let lId;
+    let rId;
+    for (let stat of stats.values()) {
+      if (stat.type == "candidate-pair" && stat.selected) {
+        lId = stat.localCandidateId;
+        rId = stat.remoteCandidateId;
+        break;
+      }
+    }
+    isnot(
+      lId,
+      undefined,
+      "Got local candidate ID " + lId + " for selected pair"
+    );
+    isnot(
+      rId,
+      undefined,
+      "Got remote candidate ID " + rId + " for selected pair"
+    );
+    let lCand = stats.get(lId);
+    let rCand = stats.get(rId);
+    if (!lCand || !rCand) {
+      ok(
+        false,
+        "failed to find candidatepair IDs or stats for local: " +
+          lId +
+          " remote: " +
+          rId
+      );
+      return;
+    }
+
+    info(
+      "checkStatsIceConnectionType verifying: local=" +
+        JSON.stringify(lCand) +
+        " remote=" +
+        JSON.stringify(rCand)
+    );
+    expectedLocalCandidateType = expectedLocalCandidateType || "host";
+    var candidateType = lCand.candidateType;
+    if (lCand.relayProtocol === "tcp" && candidateType === "relay") {
+      candidateType = "relay-tcp";
+    }
+
+    if (lCand.relayProtocol === "tls" && candidateType === "relay") {
+      candidateType = "relay-tls";
+    }
+
+    if (expectedLocalCandidateType === "srflx" && candidateType === "prflx") {
+      // Be forgiving of prflx when expecting srflx, since that can happen due
+      // to timing.
+      candidateType = "srflx";
+    }
+
+    is(
+      candidateType,
+      expectedLocalCandidateType,
+      "Local candidate type is what we expected for selected pair"
+    );
+  },
+
+  /**
+   * Compares amount of established ICE connection according to ICE candidate
+   * pairs in the stats reporting with the expected amount of connection based
+   * on the constraints.
+   *
+   * @param {object} stats
+   *        The stats to check for ICE candidate pairs
+   * @param {object} testOptions
+   *        The test options object from the PeerConnectionTest
+   */
+  checkStatsIceConnections(stats, testOptions) {
+    var numIceConnections = 0;
+    stats.forEach(stat => {
+      if (stat.type === "candidate-pair" && stat.selected) {
+        numIceConnections += 1;
+      }
+    });
+    info("ICE connections according to stats: " + numIceConnections);
+    isnot(
+      numIceConnections,
+      0,
+      "Number of ICE connections according to stats is not zero"
+    );
+    if (testOptions.bundle) {
+      if (testOptions.rtcpmux) {
+        is(numIceConnections, 1, "stats reports exactly 1 ICE connection");
+      } else {
+        is(
+          numIceConnections,
+          2,
+          "stats report exactly 2 ICE connections for media and RTCP"
+        );
+      }
+    } else {
+      var numAudioTransceivers = this._pc
+        .getTransceivers()
+        .filter(transceiver => {
+          return (
+            !transceiver.stopped && transceiver.receiver.track.kind == "audio"
+          );
+        }).length;
+
+      var numVideoTransceivers = this._pc
+        .getTransceivers()
+        .filter(transceiver => {
+          return (
+            !transceiver.stopped && transceiver.receiver.track.kind == "video"
+          );
+        }).length;
+
+      var numExpectedTransports = numAudioTransceivers + numVideoTransceivers;
+      if (!testOptions.rtcpmux) {
+        numExpectedTransports *= 2;
+      }
+
+      if (this.dataChannels.length) {
+        ++numExpectedTransports;
+      }
+
+      info(
+        "expected audio + video + data transports: " + numExpectedTransports
+      );
+      is(
+        numIceConnections,
+        numExpectedTransports,
+        "stats ICE connections matches expected A/V transports"
+      );
+    }
+  },
+
+  expectNegotiationNeeded() {
+    if (!this.observedNegotiationNeeded) {
+      this.observedNegotiationNeeded = new Promise(resolve => {
+        this.onnegotiationneeded = resolve;
+      });
+    }
+  },
+
+  /**
+   * Property-matching function for finding a certain stat in passed-in stats
+   *
+   * @param {object} stats
+   *        The stats to check from this PeerConnectionWrapper
+   * @param {object} props
+   *        The properties to look for
+   * @returns {boolean} Whether an entry containing all match-props was found.
+   */
+  hasStat(stats, props) {
+    for (let res of stats.values()) {
+      var match = true;
+      for (let prop in props) {
+        if (res[prop] !== props[prop]) {
+          match = false;
+          break;
+        }
+      }
+      if (match) {
+        return true;
+      }
+    }
+    return false;
+  },
+
+  /**
+   * Closes the connection
+   */
+  close() {
+    this._pc.close();
+    this.localMediaElements.forEach(e => e.pause());
+    info(this + ": Closed connection.");
+  },
+
+  /**
+   * Returns the string representation of the class
+   *
+   * @returns {String} The string representation
+   */
+  toString() {
+    return "PeerConnectionWrapper (" + this.label + ")";
+  },
+};
+
+// haxx to prevent SimpleTest from failing at window.onload
+function addLoadEvent() {}
+
+function loadScript(...scripts) {
+  return Promise.all(
+    scripts.map(script => {
+      var el = document.createElement("script");
+      if (typeof scriptRelativePath === "string" && script.charAt(0) !== "/") {
+        script = scriptRelativePath + script;
+      }
+      el.src = script;
+      document.head.appendChild(el);
+      return new Promise(r => {
+        el.onload = r;
+        el.onerror = r;
+      });
+    })
+  );
+}
+
+// Ensure SimpleTest.js is loaded before other scripts.
+/* import-globals-from /testing/mochitest/tests/SimpleTest/SimpleTest.js */
+/* import-globals-from head.js */
+/* import-globals-from templates.js */
+/* import-globals-from turnConfig.js */
+/* import-globals-from dataChannel.js */
+/* import-globals-from network.js */
+/* import-globals-from sdpUtils.js */
+
+var scriptsReady = loadScript("/tests/SimpleTest/SimpleTest.js").then(() => {
+  return loadScript(
+    "head.js",
+    "templates.js",
+    "turnConfig.js",
+    "dataChannel.js",
+    "network.js",
+    "sdpUtils.js"
+  );
+});
+
+function createHTML(options) {
+  return scriptsReady.then(() => realCreateHTML(options));
+}
+
+var iceServerWebsocket;
+var iceServersArray = [];
+
+var addTurnsSelfsignedCerts = () => {
+  var gUrl = SimpleTest.getTestFileURL("addTurnsSelfsignedCert.js");
+  var gScript = SpecialPowers.loadChromeScript(gUrl);
+  var certs = [];
+  // If the ICE server is running TURNS, and includes a "cert" attribute in
+  // its JSON, we set up an override that will forgive things like
+  // self-signed for it.
+  iceServersArray.forEach(iceServer => {
+    if (iceServer.hasOwnProperty("cert")) {
+      iceServer.urls.forEach(url => {
+        if (url.startsWith("turns:")) {
+          // Assumes no port or params!
+          certs.push({ cert: iceServer.cert, hostname: url.substr(6) });
+        }
+      });
+    }
+  });
+
+  return new Promise((resolve, reject) => {
+    gScript.addMessageListener("certs-added", () => {
+      resolve();
+    });
+
+    gScript.sendAsyncMessage("add-turns-certs", certs);
+  });
+};
+
+var setupIceServerConfig = useIceServer => {
+  // We disable ICE support for HTTP proxy when using a TURN server, because
+  // mochitest uses a fake HTTP proxy to serve content, which will eat our STUN
+  // packets for TURN TCP.
+  var enableHttpProxy = enable =>
+    SpecialPowers.pushPrefEnv({
+      set: [["media.peerconnection.disable_http_proxy", !enable]],
+    });
+
+  var spawnIceServer = () =>
+    new Promise((resolve, reject) => {
+      iceServerWebsocket = new WebSocket("ws://localhost:8191/");
+      iceServerWebsocket.onopen = event => {
+        info("websocket/process bridge open, starting ICE Server...");
+        iceServerWebsocket.send("iceserver");
+      };
+
+      iceServerWebsocket.onmessage = event => {
+        // The first message will contain the iceServers configuration, subsequent
+        // messages are just logging.
+        info("ICE Server: " + event.data);
+        resolve(event.data);
+      };
+
+      iceServerWebsocket.onerror = () => {
+        reject("ICE Server error: Is the ICE server websocket up?");
+      };
+
+      iceServerWebsocket.onclose = () => {
+        info("ICE Server websocket closed");
+        reject("ICE Server gone before getting configuration");
+      };
+    });
+
+  if (!useIceServer) {
+    info("Skipping ICE Server for this test");
+    return enableHttpProxy(true);
+  }
+
+  return enableHttpProxy(false)
+    .then(spawnIceServer)
+    .then(iceServersStr => {
+      iceServersArray = JSON.parse(iceServersStr);
+    })
+    .then(addTurnsSelfsignedCerts);
+};
+
+async function runNetworkTest(testFunction, fixtureOptions = {}) {
+  let { AppConstants } = SpecialPowers.ChromeUtils.importESModule(
+    "resource://gre/modules/AppConstants.sys.mjs"
+  );
+
+  await scriptsReady;
+  await runTestWhenReady(async options => {
+    await startNetworkAndTest();
+    await setupIceServerConfig(fixtureOptions.useIceServer);
+    await testFunction(options);
+    await networkTestFinished();
+  });
+}
diff --git a/dom/media/webrtc/tests/mochitests/peerconnection_audio_forced_sample_rate.js b/dom/media/webrtc/tests/mochitests/peerconnection_audio_forced_sample_rate.js
new file mode 100644
index 0000000000..d0c647be0d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/peerconnection_audio_forced_sample_rate.js
@@ -0,0 +1,32 @@
+// This function takes a sample-rate, and tests that audio flows correctly when
+// the sampling-rate at which the MTG runs is not one of the sampling-rates that
+// the MediaPipeline can work with.
+// It is in a separate file because we have an MTG per document, and we want to
+// test multiple sample-rates, so we include it in multiple HTML mochitest
+// files.
+async function test_peerconnection_audio_forced_sample_rate(forcedSampleRate) {
+  await scriptsReady;
+  await pushPrefs(["media.cubeb.force_sample_rate", forcedSampleRate]);
+  await runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    const ac = new AudioContext();
+    test.setMediaConstraints([{ audio: true }], []);
+    test.chain.replace("PC_LOCAL_GUM", [
+      function PC_LOCAL_WEBAUDIO_SOURCE(test) {
+        const oscillator = ac.createOscillator();
+        oscillator.type = "sine";
+        oscillator.frequency.value = 700;
+        oscillator.start();
+        const dest = ac.createMediaStreamDestination();
+        oscillator.connect(dest);
+        test.pcLocal.attachLocalStream(dest.stream);
+      },
+    ]);
+    test.chain.append([
+      function CHECK_REMOTE_AUDIO_FLOW(test) {
+        return test.pcRemote.checkReceivingToneFrom(ac, test.pcLocal);
+      },
+    ]);
+    return test.run();
+  });
+}
diff --git a/dom/media/webrtc/tests/mochitests/sdpUtils.js b/dom/media/webrtc/tests/mochitests/sdpUtils.js
new file mode 100644
index 0000000000..51cae10dba
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/sdpUtils.js
@@ -0,0 +1,398 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+var sdputils = {
+  // Finds the codec id / payload type given a codec format
+  // (e.g., "VP8", "VP9/90000"). `offset` tells us which one to use in case of
+  // multiple matches.
+  findCodecId(sdp, format, offset = 0) {
+    let regex = new RegExp("rtpmap:([0-9]+) " + format, "gi");
+    let match;
+    for (let i = 0; i <= offset; ++i) {
+      match = regex.exec(sdp);
+      if (!match) {
+        throw new Error(
+          "Couldn't find offset " +
+            i +
+            " of codec " +
+            format +
+            " while looking for offset " +
+            offset +
+            " in sdp:\n" +
+            sdp
+        );
+      }
+    }
+    // match[0] is the full matched string
+    // match[1] is the first parenthesis group
+    return match[1];
+  },
+
+  // Returns a list of all payload types, excluding rtx, in an sdp.
+  getPayloadTypes(sdp) {
+    const regex = /^a=rtpmap:([0-9]+) (?:(?!rtx).)*$/gim;
+    const pts = [];
+    for (const [line, pt] of sdp.matchAll(regex)) {
+      pts.push(pt);
+    }
+    return pts;
+  },
+
+  // Finds all the extmap ids in the given sdp.  Note that this does NOT
+  // consider m-sections, so a more generic version would need to
+  // look at each m-section separately.
+  findExtmapIds(sdp) {
+    var sdpExtmapIds = [];
+    extmapRegEx = /^a=extmap:([0-9+])/gm;
+    // must call exec on the regex to get each match in the string
+    while ((searchResults = extmapRegEx.exec(sdp)) !== null) {
+      // returned array has the matched text as the first item,
+      // and then one item for each capturing parenthesis that
+      // matched containing the text that was captured.
+      sdpExtmapIds.push(searchResults[1]);
+    }
+    return sdpExtmapIds;
+  },
+
+  findExtmapIdsUrnsDirections(sdp) {
+    var sdpExtmap = [];
+    extmapRegEx = /^a=extmap:([0-9+])([A-Za-z/]*) ([A-Za-z0-9_:\-\/\.]+)/gm;
+    // must call exec on the regex to get each match in the string
+    while ((searchResults = extmapRegEx.exec(sdp)) !== null) {
+      // returned array has the matched text as the first item,
+      // and then one item for each capturing parenthesis that
+      // matched containing the text that was captured.
+      var idUrn = [];
+      idUrn.push(searchResults[1]);
+      idUrn.push(searchResults[3]);
+      idUrn.push(searchResults[2].slice(1));
+      sdpExtmap.push(idUrn);
+    }
+    return sdpExtmap;
+  },
+
+  verify_unique_extmap_ids(sdp) {
+    const sdpExtmapIds = sdputils.findExtmapIdsUrnsDirections(sdp);
+
+    return sdpExtmapIds.reduce(function (result, item, index) {
+      const [id, urn, dir] = item;
+      ok(
+        !(id in result) || (result[id][0] === urn && result[id][1] === dir),
+        "ID " + id + " is unique ID for " + urn + " and direction " + dir
+      );
+      result[id] = [urn, dir];
+      return result;
+    }, {});
+  },
+
+  getMSections(sdp) {
+    return sdp
+      .split(new RegExp("^m=", "gm"))
+      .slice(1)
+      .map(s => "m=" + s);
+  },
+
+  getAudioMSections(sdp) {
+    return this.getMSections(sdp).filter(section =>
+      section.startsWith("m=audio")
+    );
+  },
+
+  getVideoMSections(sdp) {
+    return this.getMSections(sdp).filter(section =>
+      section.startsWith("m=video")
+    );
+  },
+
+  checkSdpAfterEndOfTrickle(description, testOptions, label) {
+    info("EOC-SDP: " + JSON.stringify(description));
+
+    const checkForTransportAttributes = msection => {
+      info("Checking msection: " + msection);
+      ok(
+        msection.includes("a=end-of-candidates"),
+        label + ": SDP contains end-of-candidates"
+      );
+
+      if (!msection.startsWith("m=application")) {
+        if (testOptions.rtcpmux) {
+          ok(
+            msection.includes("a=rtcp-mux"),
+            label + ": SDP contains rtcp-mux"
+          );
+        } else {
+          ok(msection.includes("a=rtcp:"), label + ": SDP contains rtcp port");
+        }
+      }
+    };
+
+    const hasOwnTransport = msection => {
+      const port0Check = new RegExp(/^m=\S+ 0 /).exec(msection);
+      if (port0Check) {
+        return false;
+      }
+      const midMatch = new RegExp(/\r\na=mid:(\S+)/).exec(msection);
+      if (!midMatch) {
+        return true;
+      }
+      const mid = midMatch[1];
+      const bundleGroupMatch = new RegExp(
+        "\\r\\na=group:BUNDLE \\S.* " + mid + "\\s+"
+      ).exec(description.sdp);
+      return bundleGroupMatch == null;
+    };
+
+    const msectionsWithOwnTransports = this.getMSections(
+      description.sdp
+    ).filter(hasOwnTransport);
+
+    ok(
+      msectionsWithOwnTransports.length,
+      "SDP should contain at least one msection with a transport"
+    );
+    msectionsWithOwnTransports.forEach(checkForTransportAttributes);
+
+    if (testOptions.ssrc) {
+      ok(description.sdp.includes("a=ssrc"), label + ": SDP contains a=ssrc");
+    } else {
+      ok(
+        !description.sdp.includes("a=ssrc"),
+        label + ": SDP does not contain a=ssrc"
+      );
+    }
+  },
+
+  // Note, we don't bother removing the fmtp lines, which makes a good test
+  // for some SDP parsing issues.
+  removeCodec(sdp, codec) {
+    var updated_sdp = sdp.replace(
+      new RegExp("a=rtpmap:" + codec + ".*\\/90000\\r\\n", ""),
+      ""
+    );
+    updated_sdp = updated_sdp.replace(
+      new RegExp("(RTP\\/SAVPF.*)( " + codec + ")(.*\\r\\n)", ""),
+      "$1$3"
+    );
+    updated_sdp = updated_sdp.replace(
+      new RegExp("a=rtcp-fb:" + codec + " nack\\r\\n", ""),
+      ""
+    );
+    updated_sdp = updated_sdp.replace(
+      new RegExp("a=rtcp-fb:" + codec + " nack pli\\r\\n", ""),
+      ""
+    );
+    updated_sdp = updated_sdp.replace(
+      new RegExp("a=rtcp-fb:" + codec + " ccm fir\\r\\n", ""),
+      ""
+    );
+    return updated_sdp;
+  },
+
+  removeAllButPayloadType(sdp, pt) {
+    return sdp.replace(
+      new RegExp("m=(\\w+ \\w+) UDP/TLS/RTP/SAVPF .*" + pt + ".*\\r\\n", "gi"),
+      "m=$1 UDP/TLS/RTP/SAVPF " + pt + "\r\n"
+    );
+  },
+
+  removeRtpMapForPayloadType(sdp, pt) {
+    return sdp.replace(new RegExp("a=rtpmap:" + pt + ".*\\r\\n", "gi"), "");
+  },
+
+  removeRtcpMux(sdp) {
+    return sdp.replace(/a=rtcp-mux\r\n/g, "");
+  },
+
+  removeSSRCs(sdp) {
+    return sdp.replace(/a=ssrc.*\r\n/g, "");
+  },
+
+  removeBundle(sdp) {
+    return sdp.replace(/a=group:BUNDLE .*\r\n/g, "");
+  },
+
+  reduceAudioMLineToPcmuPcma(sdp) {
+    return sdp.replace(
+      /m=audio .*\r\n/g,
+      "m=audio 9 UDP/TLS/RTP/SAVPF 0 8\r\n"
+    );
+  },
+
+  setAllMsectionsInactive(sdp) {
+    return sdp
+      .replace(/\r\na=sendrecv/g, "\r\na=inactive")
+      .replace(/\r\na=sendonly/g, "\r\na=inactive")
+      .replace(/\r\na=recvonly/g, "\r\na=inactive");
+  },
+
+  removeAllRtpMaps(sdp) {
+    return sdp.replace(/a=rtpmap:.*\r\n/g, "");
+  },
+
+  reduceAudioMLineToDynamicPtAndOpus(sdp) {
+    return sdp.replace(
+      /m=audio .*\r\n/g,
+      "m=audio 9 UDP/TLS/RTP/SAVPF 101 109\r\n"
+    );
+  },
+
+  addTiasBps(sdp, bps) {
+    return sdp.replace(/c=IN (.*)\r\n/g, "c=IN $1\r\nb=TIAS:" + bps + "\r\n");
+  },
+
+  removeSimulcastProperties(sdp) {
+    return sdp
+      .replace(/a=simulcast:.*\r\n/g, "")
+      .replace(/a=rid:.*\r\n/g, "")
+      .replace(
+        /a=extmap:[^\s]* urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id.*\r\n/g,
+        ""
+      )
+      .replace(
+        /a=extmap:[^\s]* urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id.*\r\n/g,
+        ""
+      );
+  },
+
+  transferSimulcastProperties(offer_sdp, answer_sdp) {
+    if (!offer_sdp.includes("a=simulcast:")) {
+      return answer_sdp;
+    }
+    ok(
+      offer_sdp.includes("a=simulcast:send "),
+      "Offer contains simulcast attribute"
+    );
+    var o_simul = offer_sdp.match(/simulcast:send (.*)([\n$])*/i);
+    var new_answer_sdp = answer_sdp + "a=simulcast:recv " + o_simul[1] + "\r\n";
+    ok(offer_sdp.includes("a=rid:"), "Offer contains RID attribute");
+    var o_rids = offer_sdp.match(/a=rid:(.*)/gi);
+    o_rids.forEach(o_rid => {
+      new_answer_sdp = new_answer_sdp + o_rid.replace(/send/, "recv") + "\r\n";
+    });
+    var extmap_id = offer_sdp.match(
+      "a=extmap:([0-9+])/sendonly urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id"
+    );
+    ok(extmap_id != null, "Offer contains RID RTP header extension");
+    new_answer_sdp =
+      new_answer_sdp +
+      "a=extmap:" +
+      extmap_id[1] +
+      "/recvonly urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\n";
+    var extmap_id = offer_sdp.match(
+      "a=extmap:([0-9+])/sendonly urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"
+    );
+    if (extmap_id != null) {
+      new_answer_sdp =
+        new_answer_sdp +
+        "a=extmap:" +
+        extmap_id[1] +
+        "/recvonly urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\n";
+    }
+
+    return new_answer_sdp;
+  },
+
+  verifySdp(
+    desc,
+    expectedType,
+    offerConstraintsList,
+    offerOptions,
+    testOptions
+  ) {
+    info("Examining this SessionDescription: " + JSON.stringify(desc));
+    info("offerConstraintsList: " + JSON.stringify(offerConstraintsList));
+    info("offerOptions: " + JSON.stringify(offerOptions));
+    ok(desc, "SessionDescription is not null");
+    is(desc.type, expectedType, "SessionDescription type is " + expectedType);
+    ok(desc.sdp.length > 10, "SessionDescription body length is plausible");
+    ok(desc.sdp.includes("a=ice-ufrag"), "ICE username is present in SDP");
+    ok(desc.sdp.includes("a=ice-pwd"), "ICE password is present in SDP");
+    ok(desc.sdp.includes("a=fingerprint"), "ICE fingerprint is present in SDP");
+    //TODO: update this for loopback support bug 1027350
+    ok(
+      !desc.sdp.includes(LOOPBACK_ADDR),
+      "loopback interface is absent from SDP"
+    );
+    var requiresTrickleIce = !desc.sdp.includes("a=candidate");
+    if (requiresTrickleIce) {
+      info("No ICE candidate in SDP -> requiring trickle ICE");
+    } else {
+      info("at least one ICE candidate is present in SDP");
+    }
+
+    //TODO: how can we check for absence/presence of m=application?
+
+    var audioTracks =
+      sdputils.countTracksInConstraint("audio", offerConstraintsList) ||
+      (offerOptions && offerOptions.offerToReceiveAudio ? 1 : 0);
+
+    info("expected audio tracks: " + audioTracks);
+    if (audioTracks == 0) {
+      ok(!desc.sdp.includes("m=audio"), "audio m-line is absent from SDP");
+    } else {
+      ok(desc.sdp.includes("m=audio"), "audio m-line is present in SDP");
+      is(
+        testOptions.opus,
+        desc.sdp.includes("a=rtpmap:109 opus/48000/2"),
+        "OPUS codec is present in SDP"
+      );
+      //TODO: ideally the rtcp-mux should be for the m=audio, and not just
+      //      anywhere in the SDP (JS SDP parser bug 1045429)
+      is(
+        testOptions.rtcpmux,
+        desc.sdp.includes("a=rtcp-mux"),
+        "RTCP Mux is offered in SDP"
+      );
+    }
+
+    var videoTracks =
+      sdputils.countTracksInConstraint("video", offerConstraintsList) ||
+      (offerOptions && offerOptions.offerToReceiveVideo ? 1 : 0);
+
+    info("expected video tracks: " + videoTracks);
+    if (videoTracks == 0) {
+      ok(!desc.sdp.includes("m=video"), "video m-line is absent from SDP");
+    } else {
+      ok(desc.sdp.includes("m=video"), "video m-line is present in SDP");
+      if (testOptions.h264) {
+        ok(
+          desc.sdp.includes("a=rtpmap:126 H264/90000") ||
+            desc.sdp.includes("a=rtpmap:97 H264/90000"),
+          "H.264 codec is present in SDP"
+        );
+      } else {
+        ok(
+          desc.sdp.includes("a=rtpmap:120 VP8/90000") ||
+            desc.sdp.includes("a=rtpmap:121 VP9/90000"),
+          "VP8 or VP9 codec is present in SDP"
+        );
+      }
+      is(
+        testOptions.rtcpmux,
+        desc.sdp.includes("a=rtcp-mux"),
+        "RTCP Mux is offered in SDP"
+      );
+      is(
+        testOptions.ssrc,
+        desc.sdp.includes("a=ssrc"),
+        "a=ssrc signaled in SDP"
+      );
+    }
+
+    return requiresTrickleIce;
+  },
+
+  /**
+   * Counts the amount of audio tracks in a given media constraint.
+   *
+   * @param constraints
+   *        The contraint to be examined.
+   */
+  countTracksInConstraint(type, constraints) {
+    if (!Array.isArray(constraints)) {
+      return 0;
+    }
+    return constraints.reduce((sum, c) => sum + (c[type] ? 1 : 0), 0);
+  },
+};
diff --git a/dom/media/webrtc/tests/mochitests/simulcast.js b/dom/media/webrtc/tests/mochitests/simulcast.js
new file mode 100644
index 0000000000..0af36478c4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/simulcast.js
@@ -0,0 +1,232 @@
+"use strict";
+/* Helper functions to munge SDP and split the sending track into
+ * separate tracks on the receiving end. This can be done in a number
+ * of ways, the one used here uses the fact that the MID and RID header
+ * extensions which are used for packet routing share the same wire
+ * format. The receiver interprets the rids from the sender as mids
+ * which allows receiving the different spatial resolutions on separate
+ * m-lines and tracks.
+ */
+
+// Borrowed from wpt, with some dependencies removed.
+
+const ridExtensions = [
+  "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id",
+  "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id",
+];
+
+function ridToMid(description, rids) {
+  const sections = SDPUtils.splitSections(description.sdp);
+  const dtls = SDPUtils.getDtlsParameters(sections[1], sections[0]);
+  const ice = SDPUtils.getIceParameters(sections[1], sections[0]);
+  const rtpParameters = SDPUtils.parseRtpParameters(sections[1]);
+  const setupValue = description.sdp.match(/a=setup:(.*)/)[1];
+  const directionValue =
+    description.sdp.match(/a=sendrecv|a=sendonly|a=recvonly|a=inactive/) ||
+    "a=sendrecv";
+  const mline = SDPUtils.parseMLine(sections[1]);
+
+  // Skip mid extension; we are replacing it with the rid extmap
+  rtpParameters.headerExtensions = rtpParameters.headerExtensions.filter(
+    ext => ext.uri != "urn:ietf:params:rtp-hdrext:sdes:mid"
+  );
+
+  for (const ext of rtpParameters.headerExtensions) {
+    if (ext.uri == "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id") {
+      ext.uri = "urn:ietf:params:rtp-hdrext:sdes:mid";
+    }
+  }
+
+  // Filter rtx as we have no way to (re)interpret rrid.
+  // Not doing this makes probing use RTX, it's not understood and ramp-up is slower.
+  rtpParameters.codecs = rtpParameters.codecs.filter(
+    c => c.name.toUpperCase() !== "RTX"
+  );
+
+  if (!rids) {
+    rids = Array.from(description.sdp.matchAll(/a=rid:(.*) send/g)).map(
+      r => r[1]
+    );
+  }
+
+  let sdp =
+    SDPUtils.writeSessionBoilerplate() +
+    SDPUtils.writeDtlsParameters(dtls, setupValue) +
+    SDPUtils.writeIceParameters(ice) +
+    "a=group:BUNDLE " +
+    rids.join(" ") +
+    "\r\n";
+  const baseRtpDescription = SDPUtils.writeRtpDescription(
+    mline.kind,
+    rtpParameters
+  );
+  for (const rid of rids) {
+    sdp +=
+      baseRtpDescription +
+      "a=mid:" +
+      rid +
+      "\r\n" +
+      "a=msid:rid-" +
+      rid +
+      " rid-" +
+      rid +
+      "\r\n";
+    sdp += directionValue + "\r\n";
+  }
+  return sdp;
+}
+
+function midToRid(description, localDescription, rids) {
+  const sections = SDPUtils.splitSections(description.sdp);
+  const dtls = SDPUtils.getDtlsParameters(sections[1], sections[0]);
+  const ice = SDPUtils.getIceParameters(sections[1], sections[0]);
+  const rtpParameters = SDPUtils.parseRtpParameters(sections[1]);
+  const setupValue = description.sdp.match(/a=setup:(.*)/)[1];
+  const directionValue =
+    description.sdp.match(/a=sendrecv|a=sendonly|a=recvonly|a=inactive/) ||
+    "a=sendrecv";
+  const mline = SDPUtils.parseMLine(sections[1]);
+
+  // Skip rid extensions; we are replacing them with the mid extmap
+  rtpParameters.headerExtensions = rtpParameters.headerExtensions.filter(
+    ext => !ridExtensions.includes(ext.uri)
+  );
+
+  for (const ext of rtpParameters.headerExtensions) {
+    if (ext.uri == "urn:ietf:params:rtp-hdrext:sdes:mid") {
+      ext.uri = "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id";
+    }
+  }
+
+  const localMid = localDescription
+    ? SDPUtils.getMid(SDPUtils.splitSections(localDescription.sdp)[1])
+    : "0";
+
+  if (!rids) {
+    rids = [];
+    for (let i = 1; i < sections.length; i++) {
+      rids.push(SDPUtils.getMid(sections[i]));
+    }
+  }
+
+  let sdp =
+    SDPUtils.writeSessionBoilerplate() +
+    SDPUtils.writeDtlsParameters(dtls, setupValue) +
+    SDPUtils.writeIceParameters(ice) +
+    "a=group:BUNDLE " +
+    localMid +
+    "\r\n";
+  sdp += SDPUtils.writeRtpDescription(mline.kind, rtpParameters);
+  // Although we are converting mids to rids, we still need a mid.
+  // The first one will be consistent with trickle ICE candidates.
+  sdp += "a=mid:" + localMid + "\r\n";
+  sdp += directionValue + "\r\n";
+
+  for (const rid of rids) {
+    const stringrid = String(rid); // allow integers
+    const choices = stringrid.split(",");
+    choices.forEach(choice => {
+      sdp += "a=rid:" + choice + " recv\r\n";
+    });
+  }
+  if (rids.length) {
+    sdp += "a=simulcast:recv " + rids.join(";") + "\r\n";
+  }
+
+  return sdp;
+}
+
+async function doOfferToSendSimulcast(offerer, answerer) {
+  await offerer.setLocalDescription();
+
+  // Is this a renegotiation? If so, we cannot remove (or reorder!) any mids,
+  // even if some rids have been removed or reordered.
+  let mids = [];
+  if (answerer.localDescription) {
+    // Renegotiation. Mids must be the same as before, because renegotiation
+    // can never remove or reorder mids, nor can it expand the simulcast
+    // envelope.
+    mids = [...answerer.localDescription.sdp.matchAll(/a=mid:(.*)/g)].map(
+      e => e[1]
+    );
+  } else {
+    // First negotiation; the mids will be exactly the same as the rids
+    const simulcastAttr = offerer.localDescription.sdp.match(
+      /a=simulcast:send (.*)/
+    );
+    if (simulcastAttr) {
+      mids = simulcastAttr[1].split(";");
+    }
+  }
+
+  const nonSimulcastOffer = ridToMid(offerer.localDescription, mids);
+  await answerer.setRemoteDescription({
+    type: "offer",
+    sdp: nonSimulcastOffer,
+  });
+}
+
+async function doAnswerToRecvSimulcast(offerer, answerer, rids) {
+  await answerer.setLocalDescription();
+  const simulcastAnswer = midToRid(
+    answerer.localDescription,
+    offerer.localDescription,
+    rids
+  );
+  await offerer.setRemoteDescription({ type: "answer", sdp: simulcastAnswer });
+}
+
+async function doOfferToRecvSimulcast(offerer, answerer, rids) {
+  await offerer.setLocalDescription();
+  const simulcastOffer = midToRid(
+    offerer.localDescription,
+    answerer.localDescription,
+    rids
+  );
+  await answerer.setRemoteDescription({ type: "offer", sdp: simulcastOffer });
+}
+
+async function doAnswerToSendSimulcast(offerer, answerer) {
+  await answerer.setLocalDescription();
+
+  // See which mids the offerer had; it will barf if we remove or reorder them
+  const mids = [...offerer.localDescription.sdp.matchAll(/a=mid:(.*)/g)].map(
+    e => e[1]
+  );
+
+  const nonSimulcastAnswer = ridToMid(answerer.localDescription, mids);
+  await offerer.setRemoteDescription({
+    type: "answer",
+    sdp: nonSimulcastAnswer,
+  });
+}
+
+async function doOfferToSendSimulcastAndAnswer(offerer, answerer, rids) {
+  await doOfferToSendSimulcast(offerer, answerer);
+  await doAnswerToRecvSimulcast(offerer, answerer, rids);
+}
+
+async function doOfferToRecvSimulcastAndAnswer(offerer, answerer, rids) {
+  await doOfferToRecvSimulcast(offerer, answerer, rids);
+  await doAnswerToSendSimulcast(offerer, answerer);
+}
+
+// This would be useful for cases other than simulcast, but we do not use it
+// anywhere else right now, nor do we have a place for wpt-friendly helpers at
+// the moment.
+function createPlaybackElement(track) {
+  const elem = document.createElement(track.kind);
+  elem.autoplay = true;
+  elem.srcObject = new MediaStream([track]);
+  elem.id = track.id;
+  return elem;
+}
+
+async function getPlaybackWithLoadedMetadata(track) {
+  const elem = createPlaybackElement(track);
+  return new Promise(resolve => {
+    elem.addEventListener("loadedmetadata", () => {
+      resolve(elem);
+    });
+  });
+}
diff --git a/dom/media/webrtc/tests/mochitests/stats.js b/dom/media/webrtc/tests/mochitests/stats.js
new file mode 100644
index 0000000000..475f8eeca9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/stats.js
@@ -0,0 +1,1596 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+"use strict";
+
+const statsExpectedByType = {
+  "inbound-rtp": {
+    expected: [
+      "trackIdentifier",
+      "id",
+      "timestamp",
+      "type",
+      "ssrc",
+      "mediaType",
+      "kind",
+      "codecId",
+      "packetsReceived",
+      "packetsLost",
+      "packetsDiscarded",
+      "bytesReceived",
+      "jitter",
+      "lastPacketReceivedTimestamp",
+      "headerBytesReceived",
+      // Always missing from libwebrtc stats
+      // "estimatedPlayoutTimestamp",
+      "jitterBufferDelay",
+      "jitterBufferEmittedCount",
+    ],
+    optional: ["remoteId", "nackCount", "qpSum"],
+    localVideoOnly: [
+      "firCount",
+      "pliCount",
+      "framesDecoded",
+      "framesDropped",
+      "discardedPackets",
+      "framesPerSecond",
+      "frameWidth",
+      "frameHeight",
+      "framesReceived",
+      "totalDecodeTime",
+      "totalInterFrameDelay",
+      "totalProcessingDelay",
+      "totalSquaredInterFrameDelay",
+    ],
+    localAudioOnly: [
+      "totalSamplesReceived",
+      // libwebrtc doesn't seem to do FEC for video
+      "fecPacketsReceived",
+      "fecPacketsDiscarded",
+      "concealedSamples",
+      "silentConcealedSamples",
+      "concealmentEvents",
+      "insertedSamplesForDeceleration",
+      "removedSamplesForAcceleration",
+      "audioLevel",
+      "totalAudioEnergy",
+      "totalSamplesDuration",
+    ],
+    unimplemented: [
+      "mediaTrackId",
+      "transportId",
+      "associateStatsId",
+      "sliCount",
+      "packetsRepaired",
+      "fractionLost",
+      "burstPacketsLost",
+      "burstLossCount",
+      "burstDiscardCount",
+      "gapDiscardRate",
+      "gapLossRate",
+    ],
+    deprecated: ["mozRtt", "isRemote"],
+  },
+  "outbound-rtp": {
+    expected: [
+      "id",
+      "timestamp",
+      "type",
+      "ssrc",
+      "mediaType",
+      "kind",
+      "codecId",
+      "packetsSent",
+      "bytesSent",
+      "remoteId",
+      "headerBytesSent",
+      "retransmittedPacketsSent",
+      "retransmittedBytesSent",
+    ],
+    optional: ["nackCount", "qpSum"],
+    localAudioOnly: [],
+    localVideoOnly: [
+      "framesEncoded",
+      "firCount",
+      "pliCount",
+      "frameWidth",
+      "frameHeight",
+      "framesSent",
+      "hugeFramesSent",
+      "totalEncodeTime",
+      "totalEncodedBytesTarget",
+    ],
+    unimplemented: ["mediaTrackId", "transportId", "sliCount", "targetBitrate"],
+    deprecated: ["isRemote"],
+  },
+  "remote-inbound-rtp": {
+    expected: [
+      "id",
+      "timestamp",
+      "type",
+      "ssrc",
+      "mediaType",
+      "kind",
+      "codecId",
+      "packetsLost",
+      "jitter",
+      "localId",
+      "totalRoundTripTime",
+      "fractionLost",
+      "roundTripTimeMeasurements",
+    ],
+    optional: ["roundTripTime", "nackCount", "packetsReceived"],
+    unimplemented: [
+      "mediaTrackId",
+      "transportId",
+      "packetsDiscarded",
+      "associateStatsId",
+      "sliCount",
+      "packetsRepaired",
+      "burstPacketsLost",
+      "burstLossCount",
+      "burstDiscardCount",
+      "gapDiscardRate",
+      "gapLossRate",
+    ],
+    deprecated: ["mozRtt", "isRemote"],
+  },
+  "remote-outbound-rtp": {
+    expected: [
+      "id",
+      "timestamp",
+      "type",
+      "ssrc",
+      "mediaType",
+      "kind",
+      "codecId",
+      "packetsSent",
+      "bytesSent",
+      "localId",
+      "remoteTimestamp",
+    ],
+    optional: ["nackCount"],
+    unimplemented: ["mediaTrackId", "transportId", "sliCount", "targetBitrate"],
+    deprecated: ["isRemote"],
+  },
+  "media-source": {
+    expected: ["id", "timestamp", "type", "trackIdentifier", "kind"],
+    unimplemented: [
+      "audioLevel",
+      "totalAudioEnergy",
+      "totalSamplesDuration",
+      "echoReturnLoss",
+      "echoReturnLossEnhancement",
+      "droppedSamplesDuration",
+      "droppedSamplesEvents",
+      "totalCaptureDelay",
+      "totalSamplesCaptured",
+      "width",
+      "height",
+      "frames",
+      "framesPerSecond",
+    ],
+    optional: [],
+    deprecated: [],
+  },
+  csrc: { skip: true },
+  codec: {
+    expected: [
+      "timestamp",
+      "type",
+      "id",
+      "payloadType",
+      "transportId",
+      "mimeType",
+      "clockRate",
+      "sdpFmtpLine",
+    ],
+    optional: ["codecType", "channels"],
+    unimplemented: [],
+    deprecated: [],
+  },
+  "peer-connection": { skip: true },
+  "data-channel": { skip: true },
+  track: { skip: true },
+  transport: { skip: true },
+  "candidate-pair": {
+    expected: [
+      "id",
+      "timestamp",
+      "type",
+      "transportId",
+      "localCandidateId",
+      "remoteCandidateId",
+      "state",
+      "priority",
+      "nominated",
+      "writable",
+      "readable",
+      "bytesSent",
+      "bytesReceived",
+      "lastPacketSentTimestamp",
+      "lastPacketReceivedTimestamp",
+    ],
+    optional: ["selected"],
+    unimplemented: [
+      "totalRoundTripTime",
+      "currentRoundTripTime",
+      "availableOutgoingBitrate",
+      "availableIncomingBitrate",
+      "requestsReceived",
+      "requestsSent",
+      "responsesReceived",
+      "responsesSent",
+      "retransmissionsReceived",
+      "retransmissionsSent",
+      "consentRequestsSent",
+    ],
+    deprecated: [],
+  },
+  "local-candidate": {
+    expected: [
+      "id",
+      "timestamp",
+      "type",
+      "address",
+      "protocol",
+      "port",
+      "candidateType",
+      "priority",
+    ],
+    optional: ["relayProtocol", "proxied"],
+    unimplemented: ["networkType", "url", "transportId"],
+    deprecated: [
+      "candidateId",
+      "portNumber",
+      "ipAddress",
+      "componentId",
+      "mozLocalTransport",
+      "transport",
+    ],
+  },
+  "remote-candidate": {
+    expected: [
+      "id",
+      "timestamp",
+      "type",
+      "address",
+      "protocol",
+      "port",
+      "candidateType",
+      "priority",
+    ],
+    optional: ["relayProtocol", "proxied"],
+    unimplemented: ["networkType", "url", "transportId"],
+    deprecated: [
+      "candidateId",
+      "portNumber",
+      "ipAddress",
+      "componentId",
+      "mozLocalTransport",
+      "transport",
+    ],
+  },
+  certificate: { skip: true },
+};
+
+["in", "out"].forEach(pre => {
+  let s = statsExpectedByType[pre + "bound-rtp"];
+  s.optional = [...s.optional, ...s.localVideoOnly, ...s.localAudioOnly];
+});
+
+//
+//  Checks that the fields in a report conform to the expectations in
+// statExpectedByType
+//
+function checkExpectedFields(report) {
+  report.forEach(stat => {
+    let expectations = statsExpectedByType[stat.type];
+    ok(expectations, "Stats type " + stat.type + " was expected");
+    // If the type is not expected or if it is flagged for skipping continue to
+    // the next
+    if (!expectations || expectations.skip) {
+      return;
+    }
+    // Check that all required fields exist
+    expectations.expected.forEach(field => {
+      ok(
+        field in stat,
+        "Expected stat field " + stat.type + "." + field + " exists"
+      );
+    });
+    // Check that each field is either expected or optional
+    let allowed = [...expectations.expected, ...expectations.optional];
+    Object.keys(stat).forEach(field => {
+      ok(
+        allowed.includes(field),
+        "Stat field " +
+          stat.type +
+          "." +
+          field +
+          ` is allowed. ${JSON.stringify(stat)}`
+      );
+    });
+
+    //
+    // Ensure that unimplemented fields are not implemented
+    //   note: if a field is implemented it should be moved to expected or
+    //   optional.
+    //
+    expectations.unimplemented.forEach(field => {
+      ok(
+        !Object.keys(stat).includes(field),
+        "Unimplemented field " + stat.type + "." + field + " does not exist."
+      );
+    });
+
+    //
+    // Ensure that all deprecated fields are not present
+    //
+    expectations.deprecated.forEach(field => {
+      ok(
+        !Object.keys(stat).includes(field),
+        "Deprecated field " + stat.type + "." + field + " does not exist."
+      );
+    });
+  });
+}
+
+function pedanticChecks(report) {
+  // Check that report is only-maplike
+  [...report.keys()].forEach(key =>
+    is(
+      report[key],
+      undefined,
+      `Report is not dictionary like, it lacks a property for key ${key}`
+    )
+  );
+  report.forEach((statObj, mapKey) => {
+    info(`"${mapKey} = ${JSON.stringify(statObj, null, 2)}`);
+  });
+  // eslint-disable-next-line complexity
+  report.forEach((statObj, mapKey) => {
+    let tested = {};
+    // Record what fields get tested.
+    // To access a field foo without marking it as tested use stat.inner.foo
+    let stat = new Proxy(statObj, {
+      get(stat, key) {
+        if (key == "inner") {
+          return stat;
+        }
+        tested[key] = true;
+        return stat[key];
+      },
+    });
+
+    let expectations = statsExpectedByType[stat.type];
+
+    if (expectations.skip) {
+      return;
+    }
+
+    // All stats share the following attributes inherited from RTCStats
+    is(stat.id, mapKey, stat.type + ".id is the same as the report key.");
+
+    // timestamp
+    ok(stat.timestamp >= 0, stat.type + ".timestamp is not less than 0");
+    // If the timebase for the timestamp is not properly set the timestamp
+    // will appear relative to the year 1970; Bug 1495446
+    const date = new Date(stat.timestamp);
+    ok(
+      date.getFullYear() > 1970,
+      `${stat.type}.timestamp is relative to current time, date=${date}`
+    );
+    //
+    // RTCStreamStats attributes with common behavior
+    //
+    // inbound-rtp, outbound-rtp, remote-inbound-rtp, remote-outbound-rtp
+    // inherit from RTCStreamStats
+    if (
+      [
+        "inbound-rtp",
+        "outbound-rtp",
+        "remote-inbound-rtp",
+        "remote-outbound-rtp",
+      ].includes(stat.type)
+    ) {
+      const isRemote = stat.type.startsWith("remote-");
+      //
+      // Common RTCStreamStats fields
+      //
+
+      // SSRC
+      ok(stat.ssrc, stat.type + ".ssrc has a value");
+
+      // kind
+      ok(
+        ["audio", "video"].includes(stat.kind),
+        stat.type + ".kind is 'audio' or 'video'"
+      );
+
+      // mediaType, renamed to kind but remains for backward compability.
+      ok(
+        ["audio", "video"].includes(stat.mediaType),
+        stat.type + ".mediaType is 'audio' or 'video'"
+      );
+
+      ok(stat.kind == stat.mediaType, "kind equals legacy mediaType");
+
+      // codecId
+      ok(stat.codecId, `${stat.type}.codecId has a value`);
+      ok(report.has(stat.codecId), `codecId ${stat.codecId} exists in report`);
+      is(
+        report.get(stat.codecId).type,
+        "codec",
+        `codecId ${stat.codecId} in report is codec type`
+      );
+      is(
+        report.get(stat.codecId).mimeType.slice(0, 5),
+        stat.kind,
+        `codecId ${stat.codecId} in report is for a mimeType of the same ` +
+          `media type as the referencing rtp stream stat`
+      );
+
+      if (isRemote) {
+        // local id
+        if (stat.localId) {
+          ok(
+            report.has(stat.localId),
+            `localId ${stat.localId} exists in report.`
+          );
+          is(
+            report.get(stat.localId).ssrc,
+            stat.ssrc,
+            "remote ssrc and local ssrc match."
+          );
+          is(
+            report.get(stat.localId).remoteId,
+            stat.id,
+            "local object has remote object as it's own remote object."
+          );
+        }
+      } else {
+        // remote id
+        if (stat.remoteId) {
+          ok(
+            report.has(stat.remoteId),
+            `remoteId ${stat.remoteId} exists in report.`
+          );
+          is(
+            report.get(stat.remoteId).ssrc,
+            stat.ssrc,
+            "remote ssrc and local ssrc match."
+          );
+          is(
+            report.get(stat.remoteId).localId,
+            stat.id,
+            "remote object has local object as it's own local object."
+          );
+        }
+      }
+
+      // nackCount
+      if (stat.nackCount) {
+        ok(
+          stat.nackCount >= 0,
+          `${stat.type}.nackCount is sane (${stat.kind}).`
+        );
+      }
+
+      if (!isRemote && stat.inner.kind == "video") {
+        // firCount
+        ok(
+          stat.firCount >= 0 && stat.firCount < 100,
+          `${stat.type}.firCount is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.firCount}`
+        );
+
+        // pliCount
+        ok(
+          stat.pliCount >= 0 && stat.pliCount < 200,
+          `${stat.type}.pliCount is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.pliCount}`
+        );
+
+        // qpSum
+        if (stat.qpSum !== undefined) {
+          ok(
+            stat.qpSum > 0,
+            `${stat.type}.qpSum is at least 0 ` +
+              `${stat.kind} test. value=${stat.qpSum}`
+          );
+        }
+      } else {
+        is(
+          stat.qpSum,
+          undefined,
+          `${stat.type}.qpSum does not exist when stat.kind != video`
+        );
+      }
+    }
+
+    if (stat.type == "inbound-rtp") {
+      //
+      // Required fields
+      //
+
+      // trackIdentifier
+      is(typeof stat.trackIdentifier, "string");
+      isnot(stat.trackIdentifier, "");
+
+      // packetsReceived
+      ok(
+        stat.packetsReceived >= 0 && stat.packetsReceived < 10 ** 5,
+        `${stat.type}.packetsReceived is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.packetsReceived}`
+      );
+
+      // packetsDiscarded
+      ok(
+        stat.packetsDiscarded >= 0 && stat.packetsDiscarded < 100,
+        `${stat.type}.packetsDiscarded is sane number for a short test. ` +
+          `value=${stat.packetsDiscarded}`
+      );
+      // bytesReceived
+      ok(
+        stat.bytesReceived >= 0 && stat.bytesReceived < 10 ** 9, // Not a magic number, just a guess
+        `${stat.type}.bytesReceived is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.bytesReceived}`
+      );
+
+      // packetsLost
+      ok(
+        stat.packetsLost < 100,
+        `${stat.type}.packetsLost is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.packetsLost}`
+      );
+
+      // This should be much lower for audio, TODO: Bug 1330575
+      let expectedJitter = stat.kind == "video" ? 0.5 : 1;
+      // jitter
+      ok(
+        stat.jitter < expectedJitter,
+        `${stat.type}.jitter is sane number for a ${stat.kind} ` +
+          `local only test. value=${stat.jitter}`
+      );
+
+      // lastPacketReceivedTimestamp
+      ok(
+        stat.lastPacketReceivedTimestamp !== undefined,
+        `${stat.type}.lastPacketReceivedTimestamp has a value`
+      );
+
+      // headerBytesReceived
+      ok(
+        stat.headerBytesReceived >= 0 && stat.headerBytesReceived < 50000,
+        `${stat.type}.headerBytesReceived is sane for a short test. ` +
+          `value=${stat.headerBytesReceived}`
+      );
+
+      // Always missing from libwebrtc stats
+      // estimatedPlayoutTimestamp
+      // ok(
+      //   stat.estimatedPlayoutTimestamp !== undefined,
+      //   `${stat.type}.estimatedPlayoutTimestamp has a value`
+      // );
+
+      // jitterBufferEmittedCount
+      let expectedJitterBufferEmmitedCount = stat.kind == "video" ? 7 : 1000;
+      ok(
+        stat.jitterBufferEmittedCount > expectedJitterBufferEmmitedCount,
+        `${stat.type}.jitterBufferEmittedCount is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.jitterBufferEmittedCount}`
+      );
+
+      // jitterBufferDelay
+      let avgJitterBufferDelay =
+        stat.jitterBufferDelay / stat.jitterBufferEmittedCount;
+      ok(
+        avgJitterBufferDelay > 0.01 && avgJitterBufferDelay < 10,
+        `${stat.type}.jitterBufferDelay is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.jitterBufferDelay}/${stat.jitterBufferEmittedCount}=${avgJitterBufferDelay}`
+      );
+
+      //
+      // Optional fields
+      //
+
+      //
+      // Local audio only stats
+      //
+      if (stat.inner.kind != "audio") {
+        expectations.localAudioOnly.forEach(field => {
+          ok(
+            stat[field] === undefined,
+            `${stat.type} does not have field ${field}` +
+              ` when kind is not 'audio'`
+          );
+        });
+      } else {
+        expectations.localAudioOnly.forEach(field => {
+          ok(
+            stat.inner[field] !== undefined,
+            stat.type + " has field " + field + " when kind is video"
+          );
+        });
+        // totalSamplesReceived
+        ok(
+          stat.totalSamplesReceived > 1000,
+          `${stat.type}.totalSamplesReceived is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.totalSamplesReceived}`
+        );
+
+        // fecPacketsReceived
+        ok(
+          stat.fecPacketsReceived >= 0 && stat.fecPacketsReceived < 10 ** 5,
+          `${stat.type}.fecPacketsReceived is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.fecPacketsReceived}`
+        );
+
+        // fecPacketsDiscarded
+        ok(
+          stat.fecPacketsDiscarded >= 0 && stat.fecPacketsDiscarded < 100,
+          `${stat.type}.fecPacketsDiscarded is sane number for a short test. ` +
+            `value=${stat.fecPacketsDiscarded}`
+        );
+        // concealedSamples
+        ok(
+          stat.concealedSamples >= 0 &&
+            stat.concealedSamples <= stat.totalSamplesReceived,
+          `${stat.type}.concealedSamples is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.concealedSamples}`
+        );
+
+        // silentConcealedSamples
+        ok(
+          stat.silentConcealedSamples >= 0 &&
+            stat.silentConcealedSamples <= stat.concealedSamples,
+          `${stat.type}.silentConcealedSamples is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.silentConcealedSamples}`
+        );
+
+        // concealmentEvents
+        ok(
+          stat.concealmentEvents >= 0 &&
+            stat.concealmentEvents <= stat.packetsReceived,
+          `${stat.type}.concealmentEvents is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.concealmentEvents}`
+        );
+
+        // insertedSamplesForDeceleration
+        ok(
+          stat.insertedSamplesForDeceleration >= 0 &&
+            stat.insertedSamplesForDeceleration <= stat.totalSamplesReceived,
+          `${stat.type}.insertedSamplesForDeceleration is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.insertedSamplesForDeceleration}`
+        );
+
+        // removedSamplesForAcceleration
+        ok(
+          stat.removedSamplesForAcceleration >= 0 &&
+            stat.removedSamplesForAcceleration <= stat.totalSamplesReceived,
+          `${stat.type}.removedSamplesForAcceleration is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.removedSamplesForAcceleration}`
+        );
+
+        // audioLevel
+        ok(
+          stat.audioLevel >= 0 && stat.audioLevel <= 128,
+          `${stat.type}.bytesReceived is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.audioLevel}`
+        );
+
+        // totalAudioEnergy
+        ok(
+          stat.totalAudioEnergy >= 0 && stat.totalAudioEnergy <= 128,
+          `${stat.type}.totalAudioEnergy is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.totalAudioEnergy}`
+        );
+
+        // totalSamplesDuration
+        ok(
+          stat.totalSamplesDuration >= 0 && stat.totalSamplesDuration <= 300,
+          `${stat.type}.totalSamplesDuration is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.totalSamplesDuration}`
+        );
+      }
+
+      //
+      // Local video only stats
+      //
+      if (stat.inner.kind != "video") {
+        expectations.localVideoOnly.forEach(field => {
+          ok(
+            stat[field] === undefined,
+            `${stat.type} does not have field ${field}` +
+              ` when kind is not 'video'`
+          );
+        });
+      } else {
+        expectations.localVideoOnly.forEach(field => {
+          ok(
+            stat.inner[field] !== undefined,
+            stat.type + " has field " + field + " when kind is video"
+          );
+        });
+        // discardedPackets
+        ok(
+          stat.discardedPackets < 100,
+          `${stat.type}.discardedPackets is a sane number for a short test. ` +
+            `value=${stat.discardedPackets}`
+        );
+        // framesPerSecond
+        ok(
+          stat.framesPerSecond > 0 && stat.framesPerSecond < 70,
+          `${stat.type}.framesPerSecond is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesPerSecond}`
+        );
+
+        // framesDecoded
+        ok(
+          stat.framesDecoded > 0 && stat.framesDecoded < 1000000,
+          `${stat.type}.framesDecoded is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesDecoded}`
+        );
+
+        // framesDropped
+        ok(
+          stat.framesDropped >= 0 && stat.framesDropped < 100,
+          `${stat.type}.framesDropped is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesDropped}`
+        );
+
+        // frameWidth
+        ok(
+          stat.frameWidth > 0 && stat.frameWidth < 100000,
+          `${stat.type}.frameWidth is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesSent}`
+        );
+
+        // frameHeight
+        ok(
+          stat.frameHeight > 0 && stat.frameHeight < 100000,
+          `${stat.type}.frameHeight is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.frameHeight}`
+        );
+
+        // totalDecodeTime
+        ok(
+          stat.totalDecodeTime >= 0 && stat.totalDecodeTime < 300,
+          `${stat.type}.totalDecodeTime is sane for a short test. ` +
+            `value=${stat.totalDecodeTime}`
+        );
+
+        // totalProcessingDelay
+        ok(
+          stat.totalProcessingDelay < 100,
+          `${stat.type}.totalProcessingDelay is sane number for a short test ` +
+            `local only test. value=${stat.totalProcessingDelay}`
+        );
+
+        // totalInterFrameDelay
+        ok(
+          stat.totalInterFrameDelay >= 0 && stat.totalInterFrameDelay < 100,
+          `${stat.type}.totalInterFrameDelay is sane for a short test. ` +
+            `value=${stat.totalInterFrameDelay}`
+        );
+
+        // totalSquaredInterFrameDelay
+        ok(
+          stat.totalSquaredInterFrameDelay >= 0 &&
+            stat.totalSquaredInterFrameDelay < 100,
+          `${stat.type}.totalSquaredInterFrameDelay is sane for a short test. ` +
+            `value=${stat.totalSquaredInterFrameDelay}`
+        );
+
+        // framesReceived
+        ok(
+          stat.framesReceived >= 0 && stat.framesReceived < 100000,
+          `${stat.type}.framesReceived is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesReceived}`
+        );
+      }
+    } else if (stat.type == "remote-inbound-rtp") {
+      // roundTripTime
+      ok(
+        stat.roundTripTime >= 0,
+        `${stat.type}.roundTripTime is sane with` +
+          `value of: ${stat.roundTripTime} (${stat.kind})`
+      );
+      //
+      // Required fields
+      //
+
+      // packetsLost
+      ok(
+        stat.packetsLost < 100,
+        `${stat.type}.packetsLost is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.packetsLost}`
+      );
+
+      // jitter
+      ok(
+        stat.jitter >= 0,
+        `${stat.type}.jitter is sane number (${stat.kind}). ` +
+          `value=${stat.jitter}`
+      );
+
+      //
+      // Optional fields
+      //
+
+      // packetsReceived
+      if (stat.packetsReceived) {
+        ok(
+          stat.packetsReceived >= 0 && stat.packetsReceived < 10 ** 5,
+          `${stat.type}.packetsReceived is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.packetsReceived}`
+        );
+      }
+
+      // totalRoundTripTime
+      ok(
+        stat.totalRoundTripTime < 50000,
+        `${stat.type}.totalRoundTripTime is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.totalRoundTripTime}`
+      );
+
+      // fractionLost
+      ok(
+        stat.fractionLost < 0.2,
+        `${stat.type}.fractionLost is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.fractionLost}`
+      );
+
+      // roundTripTimeMeasurements
+      ok(
+        stat.roundTripTimeMeasurements >= 1 &&
+          stat.roundTripTimeMeasurements < 500,
+        `${stat.type}.roundTripTimeMeasurements is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.roundTripTimeMeasurements}`
+      );
+    } else if (stat.type == "outbound-rtp") {
+      //
+      // Required fields
+      //
+
+      // packetsSent
+      ok(
+        stat.packetsSent > 0 && stat.packetsSent < 10000,
+        `${stat.type}.packetsSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.packetsSent}`
+      );
+
+      // bytesSent
+      const audio1Min = 16000 * 60; // 128kbps
+      const video1Min = 250000 * 60; // 2Mbps
+      ok(
+        stat.bytesSent > 0 &&
+          stat.bytesSent < (stat.kind == "video" ? video1Min : audio1Min),
+        `${stat.type}.bytesSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.bytesSent}`
+      );
+
+      // headerBytesSent
+      ok(
+        stat.headerBytesSent > 0 &&
+          stat.headerBytesSent < (stat.kind == "video" ? video1Min : audio1Min),
+        `${stat.type}.headerBytesSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.headerBytesSent}`
+      );
+
+      // retransmittedPacketsSent
+      ok(
+        stat.retransmittedPacketsSent >= 0 &&
+          stat.retransmittedPacketsSent <
+            (stat.kind == "video" ? video1Min : audio1Min),
+        `${stat.type}.retransmittedPacketsSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.retransmittedPacketsSent}`
+      );
+
+      // retransmittedBytesSent
+      ok(
+        stat.retransmittedBytesSent >= 0 &&
+          stat.retransmittedBytesSent <
+            (stat.kind == "video" ? video1Min : audio1Min),
+        `${stat.type}.retransmittedBytesSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.retransmittedBytesSent}`
+      );
+
+      //
+      // Optional fields
+      //
+
+      // qpSum
+      // This is supported for all of our vpx codecs (on the encode side, see
+      // bug 1519590)
+      const mimeType = report.get(stat.codecId).mimeType;
+      if (mimeType.includes("VP")) {
+        ok(
+          stat.qpSum >= 0,
+          `${stat.type}.qpSum is a sane number (${stat.kind}) ` +
+            `for ${report.get(stat.codecId).mimeType}. value=${stat.qpSum}`
+        );
+      } else if (mimeType.includes("H264")) {
+        // OpenH264 encoder records QP so we check for either condition.
+        if (!stat.qpSum && !("qpSum" in stat)) {
+          ok(
+            !stat.qpSum && !("qpSum" in stat),
+            `${stat.type}.qpSum absent for ${report.get(stat.codecId).mimeType}`
+          );
+        } else {
+          ok(
+            stat.qpSum >= 0,
+            `${stat.type}.qpSum is a sane number (${stat.kind}) ` +
+              `for ${report.get(stat.codecId).mimeType}. value=${stat.qpSum}`
+          );
+        }
+      } else {
+        ok(
+          !stat.qpSum && !("qpSum" in stat),
+          `${stat.type}.qpSum absent for ${report.get(stat.codecId).mimeType}`
+        );
+      }
+
+      //
+      // Local video only stats
+      //
+      if (stat.inner.kind != "video") {
+        expectations.localVideoOnly.forEach(field => {
+          ok(
+            stat[field] === undefined,
+            `${stat.type} does not have field ` +
+              `${field} when kind is not 'video'`
+          );
+        });
+      } else {
+        expectations.localVideoOnly.forEach(field => {
+          ok(
+            stat.inner[field] !== undefined,
+            `${stat.type} has field ` +
+              `${field} when kind is video and isRemote is false`
+          );
+        });
+
+        // framesEncoded
+        ok(
+          stat.framesEncoded >= 0 && stat.framesEncoded < 100000,
+          `${stat.type}.framesEncoded is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesEncoded}`
+        );
+
+        // frameWidth
+        ok(
+          stat.frameWidth >= 0 && stat.frameWidth < 100000,
+          `${stat.type}.frameWidth is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.frameWidth}`
+        );
+
+        // frameHeight
+        ok(
+          stat.frameHeight >= 0 && stat.frameHeight < 100000,
+          `${stat.type}.frameHeight is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.frameHeight}`
+        );
+
+        // framesSent
+        ok(
+          stat.framesSent >= 0 && stat.framesSent < 100000,
+          `${stat.type}.framesSent is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesSent}`
+        );
+
+        // hugeFramesSent
+        ok(
+          stat.hugeFramesSent >= 0 && stat.hugeFramesSent < 100000,
+          `${stat.type}.hugeFramesSent is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.hugeFramesSent}`
+        );
+
+        // totalEncodeTime
+        ok(
+          stat.totalEncodeTime >= 0,
+          `${stat.type}.totalEncodeTime is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.totalEncodeTime}`
+        );
+
+        // totalEncodedBytesTarget
+        ok(
+          stat.totalEncodedBytesTarget > 1000,
+          `${stat.type}.totalEncodedBytesTarget is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.totalEncodedBytesTarget}`
+        );
+      }
+    } else if (stat.type == "remote-outbound-rtp") {
+      //
+      // Required fields
+      //
+
+      // packetsSent
+      ok(
+        stat.packetsSent > 0 && stat.packetsSent < 10000,
+        `${stat.type}.packetsSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.packetsSent}`
+      );
+
+      // bytesSent
+      const audio1Min = 16000 * 60; // 128kbps
+      const video1Min = 250000 * 60; // 2Mbps
+      ok(
+        stat.bytesSent > 0 &&
+          stat.bytesSent < (stat.kind == "video" ? video1Min : audio1Min),
+        `${stat.type}.bytesSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.bytesSent}`
+      );
+
+      ok(
+        stat.remoteTimestamp !== undefined,
+        `${stat.type}.remoteTimestamp ` + `is not undefined (${stat.kind})`
+      );
+      const ageSeconds = (stat.timestamp - stat.remoteTimestamp) / 1000;
+      // remoteTimestamp is exact (so it can be mapped to a packet), whereas
+      // timestamp has reduced precision. It is possible that
+      // remoteTimestamp occurs a millisecond into the future from
+      // timestamp. We also subtract half a millisecond when reducing
+      // precision on libwebrtc timestamps, to counteract the potential
+      // rounding up that libwebrtc may do since it tends to round its
+      // internal timestamps to whole milliseconds. In the worst case
+      // remoteTimestamp may therefore occur 2 milliseconds ahead of
+      // timestamp.
+      ok(
+        ageSeconds >= -0.002 && ageSeconds < 30,
+        `${stat.type}.remoteTimestamp is on the same timeline as ` +
+          `${stat.type}.timestamp, and no older than 30 seconds. ` +
+          `difference=${ageSeconds}s`
+      );
+    } else if (stat.type == "media-source") {
+      // trackIdentifier
+      is(typeof stat.trackIdentifier, "string");
+      isnot(stat.trackIdentifier, "");
+
+      // kind
+      is(typeof stat.kind, "string");
+      ok(stat.kind == "audio" || stat.kind == "video");
+    } else if (stat.type == "codec") {
+      //
+      // Required fields
+      //
+
+      // mimeType & payloadType
+      switch (stat.mimeType) {
+        case "audio/opus":
+          is(stat.payloadType, 109, "codec.payloadType for opus");
+          break;
+        case "video/VP8":
+          is(stat.payloadType, 120, "codec.payloadType for VP8");
+          break;
+        case "video/VP9":
+          is(stat.payloadType, 121, "codec.payloadType for VP9");
+          break;
+        case "video/H264":
+          ok(
+            stat.payloadType == 97 || stat.payloadType == 126,
+            `codec.payloadType for H264 was ${stat.payloadType}, exp. 97 or 126`
+          );
+          break;
+        default:
+          ok(
+            false,
+            `Unexpected codec.mimeType ${stat.mimeType} for payloadType ` +
+              `${stat.payloadType}`
+          );
+          break;
+      }
+
+      // transportId
+      // (no transport stats yet)
+      ok(stat.transportId, "codec.transportId is set");
+
+      // clockRate
+      if (stat.mimeType.startsWith("audio")) {
+        is(stat.clockRate, 48000, "codec.clockRate for audio/opus");
+      } else if (stat.mimeType.startsWith("video")) {
+        is(stat.clockRate, 90000, "codec.clockRate for video");
+      }
+
+      // sdpFmtpLine
+      // (not technically mandated by spec, but expected here)
+      ok(stat.sdpFmtpLine, "codec.sdpFmtpLine is set");
+      const opusParams = [
+        "maxplaybackrate",
+        "maxaveragebitrate",
+        "usedtx",
+        "stereo",
+        "useinbandfec",
+        "cbr",
+        "ptime",
+        "minptime",
+        "maxptime",
+      ];
+      const vpxParams = ["max-fs", "max-fr"];
+      const h264Params = [
+        "packetization-mode",
+        "level-asymmetry-allowed",
+        "profile-level-id",
+        "max-fs",
+        "max-cpb",
+        "max-dpb",
+        "max-br",
+        "max-mbps",
+      ];
+      for (const param of stat.sdpFmtpLine.split(";")) {
+        const [key, value] = param.split("=");
+        if (stat.payloadType == 109) {
+          ok(
+            opusParams.includes(key),
+            `codec.sdpFmtpLine param ${key}=${value} for opus`
+          );
+        } else if (stat.payloadType == 120 || stat.payloadType == 121) {
+          ok(
+            vpxParams.includes(key),
+            `codec.sdpFmtpLine param ${key}=${value} for VPx`
+          );
+        } else if (stat.payloadType == 97 || stat.payloadType == 126) {
+          ok(
+            h264Params.includes(key),
+            `codec.sdpFmtpLine param ${key}=${value} for H264`
+          );
+          if (key == "packetization-mode") {
+            if (stat.payloadType == 97) {
+              is(value, "0", "codec.sdpFmtpLine: H264 (97) packetization-mode");
+            } else if (stat.payloadType == 126) {
+              is(
+                value,
+                "1",
+                "codec.sdpFmtpLine: H264 (126) packetization-mode"
+              );
+            }
+          }
+          if (key == "profile-level-id") {
+            is(value, "42e01f", "codec.sdpFmtpLine: H264 profile-level-id");
+          }
+        }
+      }
+
+      //
+      // Optional fields
+      //
+
+      // codecType
+      ok(
+        !Object.keys(stat).includes("codecType") ||
+          stat.codecType == "encode" ||
+          stat.codecType == "decode",
+        "codec.codecType (${codec.codecType}) is an expected value or absent"
+      );
+      let numRecvStreams = 0;
+      let numSendStreams = 0;
+      const counts = {
+        "inbound-rtp": 0,
+        "outbound-rtp": 0,
+        "remote-inbound-rtp": 0,
+        "remote-outbound-rtp": 0,
+      };
+      const [kind] = stat.mimeType.split("/");
+      report.forEach(other => {
+        if (other.type == "inbound-rtp" && other.kind == kind) {
+          numRecvStreams += 1;
+        } else if (other.type == "outbound-rtp" && other.kind == kind) {
+          numSendStreams += 1;
+        }
+        if (other.codecId == stat.id) {
+          counts[other.type] += 1;
+        }
+      });
+      const expectedCounts = {
+        encode: {
+          "inbound-rtp": 0,
+          "outbound-rtp": numSendStreams,
+          "remote-inbound-rtp": numSendStreams,
+          "remote-outbound-rtp": 0,
+        },
+        decode: {
+          "inbound-rtp": numRecvStreams,
+          "outbound-rtp": 0,
+          "remote-inbound-rtp": 0,
+          "remote-outbound-rtp": numRecvStreams,
+        },
+        absent: {
+          "inbound-rtp": numRecvStreams,
+          "outbound-rtp": numSendStreams,
+          "remote-inbound-rtp": numSendStreams,
+          "remote-outbound-rtp": numRecvStreams,
+        },
+      };
+      // Note that the logic above assumes at most one sender and at most one
+      // receiver was used to generate this stats report. If more senders or
+      // receivers are present, they'd be referring to not only this codec stat,
+      // skewing `numSendStreams` and `numRecvStreams` above.
+      // This could be fixed when we support `senderId` and `receiverId` in
+      // RTCOutboundRtpStreamStats and RTCInboundRtpStreamStats respectively.
+      for (const [key, value] of Object.entries(counts)) {
+        is(
+          value,
+          expectedCounts[stat.codecType || "absent"][key],
+          `codec.codecType ${stat.codecType || "absent"} ref from ${key} stat`
+        );
+      }
+
+      // channels
+      if (stat.mimeType.startsWith("audio")) {
+        ok(stat.channels, "codec.channels should exist for audio");
+        if (stat.channels) {
+          if (stat.sdpFmtpLine.includes("stereo=1")) {
+            is(stat.channels, 2, "codec.channels for stereo audio");
+          } else {
+            is(stat.channels, 1, "codec.channels for mono audio");
+          }
+        }
+      } else {
+        ok(!stat.channels, "codec.channels should not exist for video");
+      }
+    } else if (stat.type == "candidate-pair") {
+      info("candidate-pair is: " + JSON.stringify(stat));
+      //
+      // Required fields
+      //
+
+      // transportId
+      ok(
+        stat.transportId,
+        `${stat.type}.transportId has a value. value=` +
+          `${stat.transportId} (${stat.kind})`
+      );
+
+      // localCandidateId
+      ok(
+        stat.localCandidateId,
+        `${stat.type}.localCandidateId has a value. value=` +
+          `${stat.localCandidateId} (${stat.kind})`
+      );
+
+      // remoteCandidateId
+      ok(
+        stat.remoteCandidateId,
+        `${stat.type}.remoteCandidateId has a value. value=` +
+          `${stat.remoteCandidateId} (${stat.kind})`
+      );
+
+      // priority
+      ok(
+        stat.priority,
+        `${stat.type}.priority has a value. value=` +
+          `${stat.priority} (${stat.kind})`
+      );
+
+      // readable
+      ok(
+        stat.readable,
+        `${stat.type}.readable is true. value=${stat.readable} ` +
+          `(${stat.kind})`
+      );
+
+      // writable
+      ok(
+        stat.writable,
+        `${stat.type}.writable is true. value=${stat.writable} ` +
+          `(${stat.kind})`
+      );
+
+      // state
+      if (
+        stat.state == "succeeded" &&
+        stat.selected !== undefined &&
+        stat.selected
+      ) {
+        info("candidate-pair state is succeeded and selected is true");
+        // nominated
+        ok(
+          stat.nominated,
+          `${stat.type}.nominated is true. value=${stat.nominated} ` +
+            `(${stat.kind})`
+        );
+
+        // bytesSent
+        ok(
+          stat.bytesSent > 1000,
+          `${stat.type}.bytesSent is a sane number (>1,000) for a short ` +
+            `${stat.kind} test. value=${stat.bytesSent}`
+        );
+
+        // bytesReceived
+        ok(
+          stat.bytesReceived > 500,
+          `${stat.type}.bytesReceived is a sane number (>500) for a short ` +
+            `${stat.kind} test. value=${stat.bytesReceived}`
+        );
+
+        // lastPacketSentTimestamp
+        ok(
+          stat.lastPacketSentTimestamp,
+          `${stat.type}.lastPacketSentTimestamp has a value. value=` +
+            `${stat.lastPacketSentTimestamp} (${stat.kind})`
+        );
+
+        // lastPacketReceivedTimestamp
+        ok(
+          stat.lastPacketReceivedTimestamp,
+          `${stat.type}.lastPacketReceivedTimestamp has a value. value=` +
+            `${stat.lastPacketReceivedTimestamp} (${stat.kind})`
+        );
+      } else {
+        info("candidate-pair is _not_ both state == succeeded and selected");
+        // nominated
+        ok(
+          stat.nominated !== undefined,
+          `${stat.type}.nominated exists. value=${stat.nominated} ` +
+            `(${stat.kind})`
+        );
+        ok(
+          stat.bytesSent !== undefined,
+          `${stat.type}.bytesSent exists. value=${stat.bytesSent} ` +
+            `(${stat.kind})`
+        );
+        ok(
+          stat.bytesReceived !== undefined,
+          `${stat.type}.bytesReceived exists. value=${stat.bytesReceived} ` +
+            `(${stat.kind})`
+        );
+        ok(
+          stat.lastPacketSentTimestamp !== undefined,
+          `${stat.type}.lastPacketSentTimestamp exists. value=` +
+            `${stat.lastPacketSentTimestamp} (${stat.kind})`
+        );
+        ok(
+          stat.lastPacketReceivedTimestamp !== undefined,
+          `${stat.type}.lastPacketReceivedTimestamp exists. value=` +
+            `${stat.lastPacketReceivedTimestamp} (${stat.kind})`
+        );
+      }
+
+      //
+      // Optional fields
+      //
+      // selected
+      ok(
+        stat.selected === undefined ||
+          (stat.state == "succeeded" && stat.selected) ||
+          !stat.selected,
+        `${stat.type}.selected is undefined, true when state is succeeded, ` +
+          `or false. value=${stat.selected} (${stat.kind})`
+      );
+    } else if (
+      stat.type == "local-candidate" ||
+      stat.type == "remote-candidate"
+    ) {
+      info(`candidate is ${JSON.stringify(stat)}`);
+
+      // address
+      ok(
+        stat.address,
+        `${stat.type} has address. value=${stat.address} ` + `(${stat.kind})`
+      );
+
+      // protocol
+      ok(
+        stat.protocol,
+        `${stat.type} has protocol. value=${stat.protocol} ` + `(${stat.kind})`
+      );
+
+      // port
+      ok(
+        stat.port >= 0,
+        `${stat.type} has port >= 0. value=${stat.port} ` + `(${stat.kind})`
+      );
+      ok(
+        stat.port <= 65535,
+        `${stat.type} has port <= 65535. value=${stat.port} ` + `(${stat.kind})`
+      );
+
+      // candidateType
+      ok(
+        stat.candidateType,
+        `${stat.type} has candidateType. value=${stat.candidateType} ` +
+          `(${stat.kind})`
+      );
+
+      // priority
+      ok(
+        stat.priority > 0 && stat.priority < 2 ** 32 - 1,
+        `${stat.type} has priority between 1 and 2^32 - 1 inc. ` +
+          `value=${stat.priority} (${stat.kind})`
+      );
+
+      // relayProtocol
+      if (stat.type == "local-candidate" && stat.candidateType == "relay") {
+        ok(
+          stat.relayProtocol,
+          `relay ${stat.type} has relayProtocol. value=${stat.relayProtocol} ` +
+            `(${stat.kind})`
+        );
+      } else {
+        is(
+          stat.relayProtocol,
+          undefined,
+          `relayProtocol is undefined for candidates that are not relay and ` +
+            `local. value=${stat.relayProtocol} (${stat.kind})`
+        );
+      }
+
+      // proxied
+      if (stat.proxied) {
+        ok(
+          stat.proxied == "proxied" || stat.proxied == "non-proxied",
+          `${stat.type} has proxied. value=${stat.proxied} (${stat.kind})`
+        );
+      }
+    }
+
+    //
+    // Ensure everything was tested
+    //
+    [...expectations.expected, ...expectations.optional].forEach(field => {
+      ok(
+        Object.keys(tested).includes(field),
+        `${stat.type}.${field} was tested.`
+      );
+    });
+  });
+}
+
+function dumpStats(stats) {
+  const dict = {};
+  for (const [k, v] of stats.entries()) {
+    dict[k] = v;
+  }
+  info(`Got stats: ${JSON.stringify(dict)}`);
+}
+
+async function waitForSyncedRtcp(pc) {
+  // Ensures that RTCP is present
+  let ensureSyncedRtcp = async () => {
+    let report = await pc.getStats();
+    for (const v of report.values()) {
+      if (v.type.endsWith("bound-rtp") && !(v.remoteId || v.localId)) {
+        info(`${v.id} is missing remoteId or localId: ${JSON.stringify(v)}`);
+        return null;
+      }
+      if (v.type == "remote-inbound-rtp" && v.roundTripTime === undefined) {
+        info(`${v.id} is missing roundTripTime: ${JSON.stringify(v)}`);
+        return null;
+      }
+    }
+    return report;
+  };
+  // Returns true if there is proof in aStats of rtcp flow for all remote stats
+  // objects, compared to baseStats.
+  const hasAllRtcpUpdated = (baseStats, stats) => {
+    let hasRtcpStats = false;
+    for (const v of stats.values()) {
+      if (v.type == "remote-outbound-rtp") {
+        hasRtcpStats = true;
+        if (!v.remoteTimestamp) {
+          // `remoteTimestamp` is 0 or not present.
+          return false;
+        }
+        if (v.remoteTimestamp <= baseStats.get(v.id)?.remoteTimestamp) {
+          // `remoteTimestamp` has not advanced further than the base stats,
+          // i.e., no new sender report has been received.
+          return false;
+        }
+      } else if (v.type == "remote-inbound-rtp") {
+        hasRtcpStats = true;
+        // The ideal thing here would be to check `reportsReceived`, but it's
+        // not yet implemented.
+        if (!v.packetsReceived) {
+          // `packetsReceived` is 0 or not present.
+          return false;
+        }
+        if (v.packetsReceived <= baseStats.get(v.id)?.packetsReceived) {
+          // `packetsReceived` has not advanced further than the base stats,
+          // i.e., no new receiver report has been received.
+          return false;
+        }
+      }
+    }
+    return hasRtcpStats;
+  };
+  let attempts = 0;
+  const baseStats = await pc.getStats();
+  // Time-units are MS
+  const waitPeriod = 100;
+  const maxTime = 20000;
+  for (let totalTime = maxTime; totalTime > 0; totalTime -= waitPeriod) {
+    try {
+      let syncedStats = await ensureSyncedRtcp();
+      if (syncedStats && hasAllRtcpUpdated(baseStats, syncedStats)) {
+        dumpStats(syncedStats);
+        return syncedStats;
+      }
+    } catch (e) {
+      info(e);
+      info(e.stack);
+      throw e;
+    }
+    attempts += 1;
+    info(`waitForSyncedRtcp: no sync on attempt ${attempts}, retrying.`);
+    await wait(waitPeriod);
+  }
+  throw Error(
+    "Waiting for synced RTCP timed out after at least " + maxTime + "ms"
+  );
+}
+
+function checkSenderStats(senderStats, streamCount) {
+  const outboundRtpReports = [];
+  const remoteInboundRtpReports = [];
+  for (const v of senderStats.values()) {
+    if (v.type == "outbound-rtp") {
+      outboundRtpReports.push(v);
+    } else if (v.type == "remote-inbound-rtp") {
+      remoteInboundRtpReports.push(v);
+    }
+  }
+  is(
+    outboundRtpReports.length,
+    streamCount,
+    `Sender with ${streamCount} simulcast streams has ${streamCount} outbound-rtp reports`
+  );
+  is(
+    remoteInboundRtpReports.length,
+    streamCount,
+    `Sender with ${streamCount} simulcast streams has ${streamCount} remote-inbound-rtp reports`
+  );
+  for (const outboundRtpReport of outboundRtpReports) {
+    is(
+      outboundRtpReports.filter(r => r.ssrc == outboundRtpReport.ssrc).length,
+      1,
+      "Simulcast send track SSRCs are distinct"
+    );
+    const remoteReports = remoteInboundRtpReports.filter(
+      r => r.id == outboundRtpReport.remoteId
+    );
+    is(
+      remoteReports.length,
+      1,
+      "Simulcast send tracks have exactly one remote counterpart"
+    );
+    const remoteInboundRtpReport = remoteReports[0];
+    is(
+      outboundRtpReport.ssrc,
+      remoteInboundRtpReport.ssrc,
+      "SSRC matches for outbound-rtp and remote-inbound-rtp"
+    );
+  }
+}
+
+function PC_LOCAL_TEST_LOCAL_STATS(test) {
+  return waitForSyncedRtcp(test.pcLocal._pc).then(stats => {
+    checkExpectedFields(stats);
+    pedanticChecks(stats);
+    return Promise.all([
+      test.pcLocal._pc.getSenders().map(async s => {
+        checkSenderStats(
+          await s.getStats(),
+          Math.max(1, s.getParameters()?.encodings?.length ?? 0)
+        );
+      }),
+    ]);
+  });
+}
+
+function PC_REMOTE_TEST_REMOTE_STATS(test) {
+  return waitForSyncedRtcp(test.pcRemote._pc).then(stats => {
+    checkExpectedFields(stats);
+    pedanticChecks(stats);
+    return Promise.all([
+      test.pcRemote._pc.getSenders().map(async s => {
+        checkSenderStats(
+          await s.getStats(),
+          s.track ? Math.max(1, s.getParameters()?.encodings?.length ?? 0) : 0
+        );
+      }),
+    ]);
+  });
+}
diff --git a/dom/media/webrtc/tests/mochitests/templates.js b/dom/media/webrtc/tests/mochitests/templates.js
new file mode 100644
index 0000000000..6b7750fd2c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/templates.js
@@ -0,0 +1,615 @@
+/**
+ * Default list of commands to execute for a PeerConnection test.
+ */
+
+const STABLE = "stable";
+const HAVE_LOCAL_OFFER = "have-local-offer";
+const HAVE_REMOTE_OFFER = "have-remote-offer";
+const CLOSED = "closed";
+
+const ICE_NEW = "new";
+const GATH_NEW = "new";
+const GATH_GATH = "gathering";
+const GATH_COMPLETE = "complete";
+
+function deltaSeconds(date1, date2) {
+  return (date2.getTime() - date1.getTime()) / 1000;
+}
+
+function dumpSdp(test) {
+  if (typeof test._local_offer !== "undefined") {
+    dump("ERROR: SDP offer: " + test._local_offer.sdp.replace(/[\r]/g, ""));
+  }
+  if (typeof test._remote_answer !== "undefined") {
+    dump("ERROR: SDP answer: " + test._remote_answer.sdp.replace(/[\r]/g, ""));
+  }
+
+  if (
+    test.pcLocal &&
+    typeof test.pcLocal._local_ice_candidates !== "undefined"
+  ) {
+    dump(
+      "pcLocal._local_ice_candidates: " +
+        JSON.stringify(test.pcLocal._local_ice_candidates) +
+        "\n"
+    );
+    dump(
+      "pcLocal._remote_ice_candidates: " +
+        JSON.stringify(test.pcLocal._remote_ice_candidates) +
+        "\n"
+    );
+    dump(
+      "pcLocal._ice_candidates_to_add: " +
+        JSON.stringify(test.pcLocal._ice_candidates_to_add) +
+        "\n"
+    );
+  }
+  if (
+    test.pcRemote &&
+    typeof test.pcRemote._local_ice_candidates !== "undefined"
+  ) {
+    dump(
+      "pcRemote._local_ice_candidates: " +
+        JSON.stringify(test.pcRemote._local_ice_candidates) +
+        "\n"
+    );
+    dump(
+      "pcRemote._remote_ice_candidates: " +
+        JSON.stringify(test.pcRemote._remote_ice_candidates) +
+        "\n"
+    );
+    dump(
+      "pcRemote._ice_candidates_to_add: " +
+        JSON.stringify(test.pcRemote._ice_candidates_to_add) +
+        "\n"
+    );
+  }
+
+  if (test.pcLocal && typeof test.pcLocal.iceConnectionLog !== "undefined") {
+    dump(
+      "pcLocal ICE connection state log: " +
+        test.pcLocal.iceConnectionLog +
+        "\n"
+    );
+  }
+  if (test.pcRemote && typeof test.pcRemote.iceConnectionLog !== "undefined") {
+    dump(
+      "pcRemote ICE connection state log: " +
+        test.pcRemote.iceConnectionLog +
+        "\n"
+    );
+  }
+
+  if (
+    test.pcLocal &&
+    test.pcRemote &&
+    typeof test.pcLocal.setRemoteDescDate !== "undefined" &&
+    typeof test.pcRemote.setLocalDescDate !== "undefined"
+  ) {
+    var delta = deltaSeconds(
+      test.pcLocal.setRemoteDescDate,
+      test.pcRemote.setLocalDescDate
+    );
+    dump(
+      "Delay between pcLocal.setRemote <-> pcRemote.setLocal: " + delta + "\n"
+    );
+  }
+  if (
+    test.pcLocal &&
+    test.pcRemote &&
+    typeof test.pcLocal.setRemoteDescDate !== "undefined" &&
+    typeof test.pcLocal.setRemoteDescStableEventDate !== "undefined"
+  ) {
+    var delta = deltaSeconds(
+      test.pcLocal.setRemoteDescDate,
+      test.pcLocal.setRemoteDescStableEventDate
+    );
+    dump(
+      "Delay between pcLocal.setRemote <-> pcLocal.signalingStateStable: " +
+        delta +
+        "\n"
+    );
+  }
+  if (
+    test.pcLocal &&
+    test.pcRemote &&
+    typeof test.pcRemote.setLocalDescDate !== "undefined" &&
+    typeof test.pcRemote.setLocalDescStableEventDate !== "undefined"
+  ) {
+    var delta = deltaSeconds(
+      test.pcRemote.setLocalDescDate,
+      test.pcRemote.setLocalDescStableEventDate
+    );
+    dump(
+      "Delay between pcRemote.setLocal <-> pcRemote.signalingStateStable: " +
+        delta +
+        "\n"
+    );
+  }
+}
+
+// We need to verify that at least one candidate has been (or will be) gathered.
+function waitForAnIceCandidate(pc) {
+  return new Promise(resolve => {
+    if (!pc.localRequiresTrickleIce || pc._local_ice_candidates.length) {
+      resolve();
+    } else {
+      // In some circumstances, especially when both PCs are on the same
+      // browser, even though we are connected, the connection can be
+      // established without receiving a single candidate from one or other
+      // peer.  So we wait for at least one...
+      pc._pc.addEventListener("icecandidate", resolve);
+    }
+  }).then(() => {
+    ok(
+      pc._local_ice_candidates.length,
+      pc + " received local trickle ICE candidates"
+    );
+    isnot(
+      pc._pc.iceGatheringState,
+      GATH_NEW,
+      pc + " ICE gathering state is not 'new'"
+    );
+  });
+}
+
+async function checkTrackStats(pc, track, outbound) {
+  const audio = track.kind == "audio";
+  const msg =
+    `${pc} stats ${outbound ? "outbound " : "inbound "}` +
+    `${audio ? "audio" : "video"} rtp track id ${track.id}`;
+  const stats = await pc.getStats(track);
+  ok(
+    pc.hasStat(stats, {
+      type: outbound ? "outbound-rtp" : "inbound-rtp",
+      kind: audio ? "audio" : "video",
+    }),
+    `${msg} - found expected stats`
+  );
+  ok(
+    !pc.hasStat(stats, {
+      type: outbound ? "inbound-rtp" : "outbound-rtp",
+    }),
+    `${msg} - did not find extra stats with wrong direction`
+  );
+  ok(
+    !pc.hasStat(stats, {
+      kind: audio ? "video" : "audio",
+    }),
+    `${msg} - did not find extra stats with wrong media type`
+  );
+}
+
+function checkAllTrackStats(pc) {
+  return Promise.all([
+    ...pc
+      .getExpectedActiveReceivers()
+      .map(({ track }) => checkTrackStats(pc, track, false)),
+    ...pc
+      .getExpectedSenders()
+      .map(({ track }) => checkTrackStats(pc, track, true)),
+  ]);
+}
+
+// Commands run once at the beginning of each test, even when performing a
+// renegotiation test.
+var commandsPeerConnectionInitial = [
+  function PC_LOCAL_SETUP_ICE_LOGGER(test) {
+    test.pcLocal.logIceConnectionState();
+  },
+
+  function PC_REMOTE_SETUP_ICE_LOGGER(test) {
+    test.pcRemote.logIceConnectionState();
+  },
+
+  function PC_LOCAL_SETUP_SIGNALING_LOGGER(test) {
+    test.pcLocal.logSignalingState();
+  },
+
+  function PC_REMOTE_SETUP_SIGNALING_LOGGER(test) {
+    test.pcRemote.logSignalingState();
+  },
+
+  function PC_LOCAL_SETUP_TRACK_HANDLER(test) {
+    test.pcLocal.setupTrackEventHandler();
+  },
+
+  function PC_REMOTE_SETUP_TRACK_HANDLER(test) {
+    test.pcRemote.setupTrackEventHandler();
+  },
+
+  function PC_LOCAL_CHECK_INITIAL_SIGNALINGSTATE(test) {
+    is(
+      test.pcLocal.signalingState,
+      STABLE,
+      "Initial local signalingState is 'stable'"
+    );
+  },
+
+  function PC_REMOTE_CHECK_INITIAL_SIGNALINGSTATE(test) {
+    is(
+      test.pcRemote.signalingState,
+      STABLE,
+      "Initial remote signalingState is 'stable'"
+    );
+  },
+
+  function PC_LOCAL_CHECK_INITIAL_ICE_STATE(test) {
+    is(
+      test.pcLocal.iceConnectionState,
+      ICE_NEW,
+      "Initial local ICE connection state is 'new'"
+    );
+  },
+
+  function PC_REMOTE_CHECK_INITIAL_ICE_STATE(test) {
+    is(
+      test.pcRemote.iceConnectionState,
+      ICE_NEW,
+      "Initial remote ICE connection state is 'new'"
+    );
+  },
+
+  function PC_LOCAL_CHECK_INITIAL_CAN_TRICKLE_SYNC(test) {
+    is(
+      test.pcLocal._pc.canTrickleIceCandidates,
+      null,
+      "Local trickle status should start out unknown"
+    );
+  },
+
+  function PC_REMOTE_CHECK_INITIAL_CAN_TRICKLE_SYNC(test) {
+    is(
+      test.pcRemote._pc.canTrickleIceCandidates,
+      null,
+      "Remote trickle status should start out unknown"
+    );
+  },
+];
+
+var commandsGetUserMedia = [
+  function PC_LOCAL_GUM(test) {
+    return test.pcLocal.getAllUserMediaAndAddStreams(test.pcLocal.constraints);
+  },
+
+  function PC_REMOTE_GUM(test) {
+    return test.pcRemote.getAllUserMediaAndAddStreams(
+      test.pcRemote.constraints
+    );
+  },
+];
+
+var commandsPeerConnectionOfferAnswer = [
+  function PC_LOCAL_SETUP_ICE_HANDLER(test) {
+    test.pcLocal.setupIceCandidateHandler(test);
+  },
+
+  function PC_REMOTE_SETUP_ICE_HANDLER(test) {
+    test.pcRemote.setupIceCandidateHandler(test);
+  },
+
+  function PC_LOCAL_CREATE_OFFER(test) {
+    return test.createOffer(test.pcLocal).then(offer => {
+      is(
+        test.pcLocal.signalingState,
+        STABLE,
+        "Local create offer does not change signaling state"
+      );
+    });
+  },
+
+  function PC_LOCAL_SET_LOCAL_DESCRIPTION(test) {
+    return test
+      .setLocalDescription(test.pcLocal, test.originalOffer, HAVE_LOCAL_OFFER)
+      .then(() => {
+        is(
+          test.pcLocal.signalingState,
+          HAVE_LOCAL_OFFER,
+          "signalingState after local setLocalDescription is 'have-local-offer'"
+        );
+      });
+  },
+
+  function PC_REMOTE_GET_OFFER(test) {
+    test._local_offer = test.originalOffer;
+    test._offer_constraints = test.pcLocal.constraints;
+    test._offer_options = test.pcLocal.offerOptions;
+    return Promise.resolve();
+  },
+
+  function PC_REMOTE_SET_REMOTE_DESCRIPTION(test) {
+    return test
+      .setRemoteDescription(test.pcRemote, test._local_offer, HAVE_REMOTE_OFFER)
+      .then(() => {
+        is(
+          test.pcRemote.signalingState,
+          HAVE_REMOTE_OFFER,
+          "signalingState after remote setRemoteDescription is 'have-remote-offer'"
+        );
+      });
+  },
+
+  function PC_REMOTE_CHECK_CAN_TRICKLE_SYNC(test) {
+    is(
+      test.pcRemote._pc.canTrickleIceCandidates,
+      true,
+      "Remote thinks that local can trickle"
+    );
+  },
+
+  function PC_LOCAL_SANE_LOCAL_SDP(test) {
+    test.pcLocal.localRequiresTrickleIce = sdputils.verifySdp(
+      test._local_offer,
+      "offer",
+      test._offer_constraints,
+      test._offer_options,
+      test.testOptions
+    );
+  },
+
+  function PC_REMOTE_SANE_REMOTE_SDP(test) {
+    test.pcRemote.remoteRequiresTrickleIce = sdputils.verifySdp(
+      test._local_offer,
+      "offer",
+      test._offer_constraints,
+      test._offer_options,
+      test.testOptions
+    );
+  },
+
+  function PC_REMOTE_CREATE_ANSWER(test) {
+    return test.createAnswer(test.pcRemote).then(answer => {
+      is(
+        test.pcRemote.signalingState,
+        HAVE_REMOTE_OFFER,
+        "Remote createAnswer does not change signaling state"
+      );
+    });
+  },
+
+  function PC_REMOTE_SET_LOCAL_DESCRIPTION(test) {
+    return test
+      .setLocalDescription(test.pcRemote, test.originalAnswer, STABLE)
+      .then(() => {
+        is(
+          test.pcRemote.signalingState,
+          STABLE,
+          "signalingState after remote setLocalDescription is 'stable'"
+        );
+      });
+  },
+
+  function PC_LOCAL_GET_ANSWER(test) {
+    test._remote_answer = test.originalAnswer;
+    test._answer_constraints = test.pcRemote.constraints;
+    return Promise.resolve();
+  },
+
+  function PC_LOCAL_SET_REMOTE_DESCRIPTION(test) {
+    return test
+      .setRemoteDescription(test.pcLocal, test._remote_answer, STABLE)
+      .then(() => {
+        is(
+          test.pcLocal.signalingState,
+          STABLE,
+          "signalingState after local setRemoteDescription is 'stable'"
+        );
+      });
+  },
+
+  function PC_REMOTE_SANE_LOCAL_SDP(test) {
+    test.pcRemote.localRequiresTrickleIce = sdputils.verifySdp(
+      test._remote_answer,
+      "answer",
+      test._offer_constraints,
+      test._offer_options,
+      test.testOptions
+    );
+  },
+  function PC_LOCAL_SANE_REMOTE_SDP(test) {
+    test.pcLocal.remoteRequiresTrickleIce = sdputils.verifySdp(
+      test._remote_answer,
+      "answer",
+      test._offer_constraints,
+      test._offer_options,
+      test.testOptions
+    );
+  },
+
+  function PC_LOCAL_CHECK_CAN_TRICKLE_SYNC(test) {
+    is(
+      test.pcLocal._pc.canTrickleIceCandidates,
+      true,
+      "Local thinks that remote can trickle"
+    );
+  },
+
+  function PC_LOCAL_WAIT_FOR_ICE_CONNECTED(test) {
+    return test.pcLocal.waitForIceConnected().then(() => {
+      info(
+        test.pcLocal +
+          ": ICE connection state log: " +
+          test.pcLocal.iceConnectionLog
+      );
+    });
+  },
+
+  function PC_REMOTE_WAIT_FOR_ICE_CONNECTED(test) {
+    return test.pcRemote.waitForIceConnected().then(() => {
+      info(
+        test.pcRemote +
+          ": ICE connection state log: " +
+          test.pcRemote.iceConnectionLog
+      );
+    });
+  },
+
+  function PC_LOCAL_VERIFY_ICE_GATHERING(test) {
+    return waitForAnIceCandidate(test.pcLocal);
+  },
+
+  function PC_REMOTE_VERIFY_ICE_GATHERING(test) {
+    return waitForAnIceCandidate(test.pcRemote);
+  },
+
+  function PC_LOCAL_WAIT_FOR_MEDIA_FLOW(test) {
+    return test.pcLocal.waitForMediaFlow();
+  },
+
+  function PC_REMOTE_WAIT_FOR_MEDIA_FLOW(test) {
+    return test.pcRemote.waitForMediaFlow();
+  },
+
+  function PC_LOCAL_CHECK_STATS(test) {
+    return test.pcLocal.getStats().then(stats => {
+      test.pcLocal.checkStats(stats);
+    });
+  },
+
+  function PC_REMOTE_CHECK_STATS(test) {
+    return test.pcRemote.getStats().then(stats => {
+      test.pcRemote.checkStats(stats);
+    });
+  },
+
+  function PC_LOCAL_CHECK_ICE_CONNECTION_TYPE(test) {
+    return test.pcLocal.getStats().then(stats => {
+      test.pcLocal.checkStatsIceConnectionType(
+        stats,
+        test.testOptions.expectedLocalCandidateType
+      );
+    });
+  },
+
+  function PC_REMOTE_CHECK_ICE_CONNECTION_TYPE(test) {
+    return test.pcRemote.getStats().then(stats => {
+      test.pcRemote.checkStatsIceConnectionType(
+        stats,
+        test.testOptions.expectedRemoteCandidateType
+      );
+    });
+  },
+
+  function PC_LOCAL_CHECK_ICE_CONNECTIONS(test) {
+    return test.pcLocal.getStats().then(stats => {
+      test.pcLocal.checkStatsIceConnections(stats, test.testOptions);
+    });
+  },
+
+  function PC_REMOTE_CHECK_ICE_CONNECTIONS(test) {
+    return test.pcRemote.getStats().then(stats => {
+      test.pcRemote.checkStatsIceConnections(stats, test.testOptions);
+    });
+  },
+
+  function PC_LOCAL_CHECK_MSID(test) {
+    return test.pcLocal.checkLocalMsids();
+  },
+  function PC_REMOTE_CHECK_MSID(test) {
+    return test.pcRemote.checkLocalMsids();
+  },
+
+  function PC_LOCAL_CHECK_TRACK_STATS(test) {
+    return checkAllTrackStats(test.pcLocal);
+  },
+  function PC_REMOTE_CHECK_TRACK_STATS(test) {
+    return checkAllTrackStats(test.pcRemote);
+  },
+  function PC_LOCAL_VERIFY_SDP_AFTER_END_OF_TRICKLE(test) {
+    if (test.pcLocal.endOfTrickleSdp) {
+      /* In case the endOfTrickleSdp promise is resolved already it will win the
+       * race because it gets evaluated first. But if endOfTrickleSdp is still
+       * pending the rejection will win the race. */
+      return Promise.race([
+        test.pcLocal.endOfTrickleSdp,
+        Promise.reject("No SDP"),
+      ]).then(
+        sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(
+            sdp,
+            test.testOptions,
+            test.pcLocal.label
+          ),
+        () =>
+          info(
+            "pcLocal: Gathering is not complete yet, skipping post-gathering SDP check"
+          )
+      );
+    }
+  },
+  function PC_REMOTE_VERIFY_SDP_AFTER_END_OF_TRICKLE(test) {
+    if (test.pcRemote.endOfTrickleSdp) {
+      /* In case the endOfTrickleSdp promise is resolved already it will win the
+       * race because it gets evaluated first. But if endOfTrickleSdp is still
+       * pending the rejection will win the race. */
+      return Promise.race([
+        test.pcRemote.endOfTrickleSdp,
+        Promise.reject("No SDP"),
+      ]).then(
+        sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(
+            sdp,
+            test.testOptions,
+            test.pcRemote.label
+          ),
+        () =>
+          info(
+            "pcRemote: Gathering is not complete yet, skipping post-gathering SDP check"
+          )
+      );
+    }
+  },
+];
+
+function PC_LOCAL_REMOVE_ALL_BUT_H264_FROM_OFFER(test) {
+  isnot(
+    test.originalOffer.sdp.search("H264/90000"),
+    -1,
+    "H.264 should be present in the SDP offer"
+  );
+  test.originalOffer.sdp = sdputils.removeCodec(
+    sdputils.removeCodec(
+      sdputils.removeCodec(test.originalOffer.sdp, 120),
+      121,
+      97
+    )
+  );
+  info("Updated H264 only offer: " + JSON.stringify(test.originalOffer));
+}
+
+function PC_LOCAL_REMOVE_BUNDLE_FROM_OFFER(test) {
+  test.originalOffer.sdp = sdputils.removeBundle(test.originalOffer.sdp);
+  info("Updated no bundle offer: " + JSON.stringify(test.originalOffer));
+}
+
+function PC_LOCAL_REMOVE_RTCPMUX_FROM_OFFER(test) {
+  test.originalOffer.sdp = sdputils.removeRtcpMux(test.originalOffer.sdp);
+  info("Updated no RTCP-Mux offer: " + JSON.stringify(test.originalOffer));
+}
+
+function PC_LOCAL_REMOVE_SSRC_FROM_OFFER(test) {
+  test.originalOffer.sdp = sdputils.removeSSRCs(test.originalOffer.sdp);
+  info("Updated no SSRCs offer: " + JSON.stringify(test.originalOffer));
+}
+
+function PC_REMOTE_REMOVE_SSRC_FROM_ANSWER(test) {
+  test.originalAnswer.sdp = sdputils.removeSSRCs(test.originalAnswer.sdp);
+  info("Updated no SSRCs answer: " + JSON.stringify(test.originalAnswer));
+}
+
+var addRenegotiation = (chain, commands, checks) => {
+  chain.append(commands);
+  chain.append(commandsPeerConnectionOfferAnswer);
+  if (checks) {
+    chain.append(checks);
+  }
+};
+
+var addRenegotiationAnswerer = (chain, commands, checks) => {
+  chain.append(function SWAP_PC_LOCAL_PC_REMOTE(test) {
+    var temp = test.pcLocal;
+    test.pcLocal = test.pcRemote;
+    test.pcRemote = temp;
+  });
+  addRenegotiation(chain, commands, checks);
+};
diff --git a/dom/media/webrtc/tests/mochitests/test_1488832.html b/dom/media/webrtc/tests/mochitests/test_1488832.html
new file mode 100644
index 0000000000..8798994b24
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_1488832.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+<script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<iframe id="testframe"></iframe>
+<script>
+"use strict";
+
+createHTML({
+  title: "gUM shutdown race",
+  bug: "1488832"
+});
+
+runTest(async () => {
+  testframe.srcdoc = `
+  <html>
+  <head>
+  <script>
+  function start() {
+    for (let i = 0; i < 16; i++) {
+      window.navigator.mediaDevices.getUserMedia({video: true})
+      setTimeout('location.reload()', 100)
+    }
+  }
+  document.addEventListener('DOMContentLoaded', start)
+  </` + `script>
+  </head>
+  </html>`;
+
+  await wait(10000);
+  testframe.srcdoc = "";
+});
+</script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_1717318.html b/dom/media/webrtc/tests/mochitests/test_1717318.html
new file mode 100644
index 0000000000..425bd29e7e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_1717318.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>PC construct with no global object (bug 1717318)</title>
+  <script type="text/javascript" src="/MochiKit/MochiKit.js"></script>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script class="testbody" type="text/javascript">
+"use strict";
+
+// nsIArray is not special here, it could be pretty much any interface.
+// We do this outside the try block just in case someday the interface is
+// removed.
+const dummyInterface = SpecialPowers.Components.interfaces.nsIArray;
+ok(dummyInterface, "nsIArray should exist");
+try {
+  // Just don't crash.
+  SpecialPowers.Components.classes["@mozilla.org/peerconnection;1"]
+      .createInstance(dummyInterface);
+} catch (e) {}
+
+</script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_a_noOp.html b/dom/media/webrtc/tests/mochitests/test_a_noOp.html
new file mode 100644
index 0000000000..971f5d7666
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_a_noOp.html
@@ -0,0 +1,32 @@
+<!DOCTYPE HTML>
+<html>
+<!--
+https://bugzilla.mozilla.org/show_bug.cgi?id=1264772
+-->
+<head>
+  <title>Test for Bug 1264772</title>
+  <script type="text/javascript" src="/MochiKit/MochiKit.js"></script>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<a target="_blank" href="https://bugzilla.mozilla.org/show_bug.cgi?id=1264772">Mozilla Bug 1264772</a>
+<p id="display"></p>
+<div id="content" style="display: none">
+</div>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+/** Test for Bug 1264772 **/
+// The WebRTC tests seem to have more problems with intermittents (at
+// least on Android) if they run first in a test run. This is a dummy test
+// to ensure that the browser is ready prior to running any actual WebRTC
+// tests.
+//
+// Note: mochitests are run in alphabetical order, so it is not sufficient
+//       for this test to appear first in the manifest.
+ok(true, "test passed");
+
+</script>
+</pre>
+</body>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudio.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudio.html
new file mode 100644
index 0000000000..06ca9562ad
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudio.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796895",
+    title: "Basic data channel audio connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideo.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideo.html
new file mode 100644
index 0000000000..ea534ca2e7
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideo.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796891",
+    title: "Basic data channel audio/video connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideoCombined.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideoCombined.html
new file mode 100644
index 0000000000..d5409986ec
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideoCombined.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796891",
+    title: "Basic data channel audio/video connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    test.setMediaConstraints([{audio: true, video: true}],
+                             [{audio: true, video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideoNoBundle.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideoNoBundle.html
new file mode 100644
index 0000000000..7dc22d86ad
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideoNoBundle.html
@@ -0,0 +1,27 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1016476",
+    title: "Basic data channel audio/video connection without bundle"
+  });
+
+var test;
+runNetworkTest(function (options) {
+  options = options || { };
+  options.bundle = false;
+  test = new PeerConnectionTest(options);
+  addInitialDataChannel(test.chain);
+  test.setMediaConstraints([{audio: true}, {video: true}],
+                           [{audio: true}, {video: true}]);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_basicDataOnly.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicDataOnly.html
new file mode 100644
index 0000000000..98e72f7a21
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicDataOnly.html
@@ -0,0 +1,24 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796894",
+    title: "Basic datachannel only connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_basicVideo.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicVideo.html
new file mode 100644
index 0000000000..90f2d7caff
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicVideo.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796889",
+    title: "Basic data channel video connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_bug1013809.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_bug1013809.html
new file mode 100644
index 0000000000..e36caebab4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_bug1013809.html
@@ -0,0 +1,27 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796895",
+    title: "Basic data channel audio connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    var sld = test.chain.remove("PC_REMOTE_SET_LOCAL_DESCRIPTION");
+    test.chain.insertAfter("PC_LOCAL_SET_REMOTE_DESCRIPTION", sld);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_dataOnlyBufferedAmountLow.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_dataOnlyBufferedAmountLow.html
new file mode 100644
index 0000000000..26767e0865
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_dataOnlyBufferedAmountLow.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1051685",
+    title: "Verify bufferedAmountLowThreshold works"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    test.chain.insertAfter('PC_REMOTE_CHECK_ICE_CONNECTIONS', commandsCheckLargeXfer);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_dtlsVersions.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_dtlsVersions.html
new file mode 100644
index 0000000000..6f0cbc5d3d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_dtlsVersions.html
@@ -0,0 +1,38 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1284103",
+    title: "Test basic data channel audio connection for supported DTLS versions"
+  });
+
+  async function testDtlsVersion(options, version) {
+    await SpecialPowers.pushPrefEnv({
+      set: [
+        ["media.peerconnection.dtls.version.min", version],
+        ["media.peerconnection.dtls.version.max", version]
+      ]
+    });
+
+    const test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+
+    await test.run();
+  }
+
+  runNetworkTest(async (options) => {
+    // 770 = DTLS 1.0, 771 = DTLS 1.2, 772 = DTLS 1.3
+    for (var version = 770; version <= 772; version++) {
+      await testDtlsVersion(options, version);
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_hostnameObfuscation.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_hostnameObfuscation.html
new file mode 100644
index 0000000000..d0790fb9c9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_hostnameObfuscation.html
@@ -0,0 +1,59 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1592620",
+    title: "Blocklist to disable hostname obfuscation"
+  });
+
+  async function testBlocklist(options, blocklistEntry, shouldBeObfuscated) {
+    let test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+
+    if (blocklistEntry !== null) {
+      await SpecialPowers.pushPrefEnv({
+        set: [
+          ["media.peerconnection.ice.obfuscate_host_addresses.blocklist",
+           blocklistEntry]
+        ]
+      });
+    }
+
+    test.chain.insertAfter('PC_LOCAL_WAIT_FOR_ICE_CONNECTED', [
+      async function CHECK_LOCAL_CANDIDATES() {
+        const stats = await test.pcLocal.getStats();
+        stats.forEach(s => {
+          if (s.type === 'local-candidate') {
+            if (shouldBeObfuscated) {
+              ok(s.address.includes(".local"), "address should be obfuscated");
+            } else {
+              ok(!s.address.includes(".local"), "address should not be obfuscated");
+            }
+          }
+        });
+      }]);
+
+    await test.run();
+  }
+
+  runNetworkTest(async (options) => {
+    await SpecialPowers.pushPrefEnv({
+      set: [["media.peerconnection.ice.obfuscate_host_addresses", true]]
+    });
+    await testBlocklist(options, null, true);
+    await testBlocklist(options, "", true);
+    await testBlocklist(options, "example.com", true);
+    await testBlocklist(options, "mochi.test", false);
+    await testBlocklist(options, "example.com,mochi.test", false);
+    await testBlocklist(options, "*.test", false);
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_noOffer.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_noOffer.html
new file mode 100644
index 0000000000..a6e9fa5214
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_noOffer.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "856319",
+    title: "Don't offer m=application unless createDataChannel is called first"
+  });
+
+  runNetworkTest(async function () {
+    const pc = new RTCPeerConnection();
+
+    // necessary to circumvent bug 864109
+    const options = { offerToReceiveAudio: true };
+
+    const errorCallback = generateErrorCallback();
+    try {
+      const offer = await pc.createOffer(options);
+      ok(!offer.sdp.includes("m=application"),
+        "m=application is not contained in the SDP");
+    } catch(e) {
+      errorCallback(e);
+    }
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_stats.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_stats.html
new file mode 100644
index 0000000000..4498e2d23a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_stats.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1218356",
+    title: "DataChannel stats"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.chain.remove('PC_LOCAL_CHECK_STATS');
+    test.chain.remove('PC_REMOTE_CHECK_STATS');
+    addInitialDataChannel(test.chain);
+    test.chain.removeAfter("PC_REMOTE_CHECK_ICE_CONNECTIONS");
+    test.chain.insertAfter("PC_REMOTE_CHECK_ICE_CONNECTIONS",
+      async function TEST_DATA_CHANNEL_STATS(test) {
+        const channel = test.pcLocal.dataChannels[0];
+        test.pcRemote.dataChannels[0].onbufferedamountlow = () => {};
+        test.pcRemote.dataChannels[0].send(`Sending Message`);
+        channel.onbufferedamountlow = () => {};
+        const event = await new Promise( r => channel.onmessage = r);
+        info(`Received message: "${event.data}"`);
+        const report = await test.pcLocal.getStats();
+        info(`Received Stats ${JSON.stringify([...report.values()], null, 2)}\n`);
+        const stats = [...report.values()].find(block => block.type == "data-channel");
+        info(`DataChannel stats ${JSON.stringify(stats, null, 2)}`);
+        is(stats.label, channel.label, 'DataChannel stats has correct label');
+        is(stats.protocol, channel.protocol,
+           'DataChannel stats has correct protocol');
+        is(stats.dataChannelIdentifier, channel.id,
+           'DataChannel stats has correct dataChannelIdentifier');
+        is(stats.state, channel.readyState, 'DataChannel has correct state');
+        is(stats.bytesReceived, 15, 'DataChannel has correct bytesReceived');
+        is(stats.bytesSent, 0, 'DataChannel has correct bytesSent');
+        is(stats.messagesReceived, 1,
+           'DataChannel has correct messagesReceived');
+        is(stats.messagesSent, 0, 'DataChannel has correct messagesSent');
+    });
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_defaultAudioConstraints.html b/dom/media/webrtc/tests/mochitests/test_defaultAudioConstraints.html
new file mode 100644
index 0000000000..8e0db48fff
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_defaultAudioConstraints.html
@@ -0,0 +1,80 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "Test that the audio constraints that observe at the audio constraints we expect.",
+  bug: "1509842"
+});
+
+runTest(async () => {
+  // We need a real device to get a MediaEngine supporting constraints
+  let audioDevice = SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (!audioDevice) {
+    todo(false, "No device set by framework. Try --use-test-media-devices");
+    return;
+  }
+
+  // Get a gUM track with the default settings, check that they are what we
+  // expect.
+  let stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+  let track = stream.getAudioTracks()[0];
+  let defaultSettings = track.getSettings();
+
+  is(defaultSettings.echoCancellation, true,
+      "Echo cancellation should be ON by default.");
+  is(defaultSettings.noiseSuppression, true,
+      "Noise suppression should be ON by default.");
+  is(defaultSettings.autoGainControl, true,
+      "Automatic gain control should be ON by default.");
+
+  track.stop();
+
+  // This is UA-dependant, and belongs in a Mochitest, not in a WPT.
+  // When a gUM track has been requested with `echoCancellation` OFF, check that
+  // `noiseSuppression` and `autoGainControl` are off as well.
+  stream =
+    await navigator.mediaDevices.getUserMedia({audio:{echoCancellation: false}});
+  track = stream.getAudioTracks()[0];
+  defaultSettings = track.getSettings();
+
+  is(defaultSettings.echoCancellation, false,
+      "Echo cancellation should be OFF when requested.");
+  is(defaultSettings.noiseSuppression, false,
+      `Noise suppression should be OFF when echoCancellation is the only
+      constraint and is OFF.`);
+  is(defaultSettings.autoGainControl, false,
+      `Automatic gain control should be OFF when echoCancellation is the only
+      constraint and is OFF.`);
+
+  track.stop();
+
+  // When a gUM track has been requested with `echoCancellation` OFF, check that
+  // `noiseSuppression` and `autoGainControl` are not OFF as well if another
+  // constraint has been specified.
+  stream =
+    await navigator.mediaDevices.getUserMedia({audio:{echoCancellation: false,
+                                                      autoGainControl: true}});
+  track = stream.getAudioTracks()[0];
+  defaultSettings = track.getSettings();
+
+  is(defaultSettings.echoCancellation, false,
+      "Echo cancellation should be OFF when requested.");
+  is(defaultSettings.noiseSuppression, false,
+      `Noise suppression should be OFF when echoCancellation is OFF and another
+      constraint has been specified.`);
+  is(defaultSettings.autoGainControl, true,
+      "Auto gain control should be ON when requested.");
+
+  track.stop();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_enumerateDevices.html b/dom/media/webrtc/tests/mochitests/test_enumerateDevices.html
new file mode 100644
index 0000000000..48bec0006a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_enumerateDevices.html
@@ -0,0 +1,141 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({ title: "Run enumerateDevices code", bug: "1046245" });
+/**
+  Tests covering enumerateDevices API and deviceId constraint. Exercise code.
+*/
+
+async function mustSucceedWithStream(msg, f) {
+  try {
+    const stream = await f();
+    for (const track of stream.getTracks()) {
+      track.stop();
+    }
+    ok(true, msg + " must succeed");
+  } catch (e) {
+    is(e.name, null, msg + " must succeed: " + e.message);
+  }
+}
+
+async function mustFailWith(msg, reason, constraint, f) {
+  try {
+    await f();
+    ok(false, msg + " must fail");
+  } catch(e) {
+    is(e.name, reason, msg + " must fail: " + e.message);
+    if (constraint) {
+      is(e.constraint, constraint, msg + " must fail w/correct constraint.");
+    }
+  }
+}
+
+const gUM = c => navigator.mediaDevices.getUserMedia(c);
+
+const kinds = ["videoinput", "audioinput", "audiooutput"];
+
+function validateDevice({kind, label, deviceId, groupId}) {
+  ok(kinds.includes(kind), "Known device kind");
+  is(deviceId.length, 44, "deviceId length id as expected for Firefox");
+  ok(label.length !== undefined, "Device label: " + label);
+  isnot(groupId, "", "groupId must be present.");
+}
+
+runTest(async () => {
+  await pushPrefs(["media.navigator.streams.fake", true]);
+
+  // Validate enumerated devices after gUM.
+  for (const track of (await gUM({video: true, audio: true})).getTracks()) {
+    track.stop();
+  }
+
+  let devices = await navigator.mediaDevices.enumerateDevices();
+  ok(devices.length, "At least one device found");
+  const jsoned = JSON.parse(JSON.stringify(devices));
+  is(jsoned[0].kind, devices[0].kind, "kind survived serializer");
+  is(jsoned[0].deviceId, devices[0].deviceId, "deviceId survived serializer");
+  for (const device of devices) {
+    validateDevice(device);
+    if (device.kind == "audiooutput") continue;
+    // Test deviceId constraint
+    let deviceId = device.deviceId;
+    let constraints = (device.kind == "videoinput") ? { video: { deviceId } }
+                                                    : { audio: { deviceId } };
+    for (const track of (await gUM(constraints)).getTracks()) {
+      is(typeof(track.label), "string", "Track label is a string");
+      is(track.label, device.label, "Track label is the device label");
+      track.stop();
+    }
+  }
+
+  const unknownId = "unknown9qHr8B0JIbcHlbl9xR+jMbZZ8WyoPfpCXPfc=";
+
+  // Check deviceId failure paths for video.
+
+  await mustSucceedWithStream("unknown plain deviceId on video",
+                              () => gUM({ video: { deviceId: unknownId } }));
+  await mustSucceedWithStream("unknown plain deviceId on audio",
+                              () => gUM({ audio: { deviceId: unknownId } }));
+  await mustFailWith("unknown exact deviceId on video",
+                     "OverconstrainedError", "deviceId",
+                     () => gUM({ video: { deviceId: { exact: unknownId } } }));
+  await mustFailWith("unknown exact deviceId on audio",
+                     "OverconstrainedError", "deviceId",
+                     () => gUM({ audio: { deviceId: { exact: unknownId } } }));
+
+  // Check that deviceIds are stable for same origin and differ across origins.
+
+  const path = "/tests/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe.html";
+  const origins = ["https://example.com", "https://test1.example.com"];
+  info(window.location);
+
+  const haveDevicesMap = new Promise(resolve => {
+    const map = new Map();
+    window.addEventListener("message", ({origin, data}) => {
+      ok(origins.includes(origin), "Got message from expected origin");
+      map.set(origin, JSON.parse(data));
+      if (map.size < origins.length) return;
+      resolve(map);
+    });
+  });
+
+  await Promise.all(origins.map(origin => {
+    const iframe = document.createElement("iframe");
+    iframe.src = origin + path;
+    iframe.allow = "camera;microphone;speaker-selection";
+    info(iframe.src);
+    document.documentElement.appendChild(iframe);
+    return new Promise(resolve => iframe.onload = resolve);
+  }));
+  let devicesMap = await haveDevicesMap;
+  let [sameOriginDevices, differentOriginDevices] = origins.map(o => devicesMap.get(o));
+
+  is(sameOriginDevices.length, devices.length, "same origin same devices");
+  is(differentOriginDevices.length, devices.length, "cross origin same devices");
+  [...sameOriginDevices, ...differentOriginDevices].forEach(d => validateDevice(d));
+
+  for (const device of sameOriginDevices) {
+    ok(devices.find(d => d.deviceId == device.deviceId),
+       "Same origin deviceId for " + device.label + " must match");
+  }
+  for (const device of differentOriginDevices) {
+    ok(!devices.find(d => d.deviceId == device.deviceId),
+         "Different origin deviceId for " + device.label + " must be different");
+  }
+
+  // Check the special case of no devices found.
+  await pushPrefs(["media.navigator.streams.fake", false],
+                  ["media.audio_loopback_dev", "none"],
+                  ["media.video_loopback_dev", "none"]);
+  devices = await navigator.mediaDevices.enumerateDevices();
+  is(devices.length, 0, "No devices");
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_enumerateDevices_getUserMediaFake.html b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_getUserMediaFake.html
new file mode 100644
index 0000000000..7952bcba1b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_getUserMediaFake.html
@@ -0,0 +1,63 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <meta charset="utf-8">
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+  <script>
+"use strict";
+
+createHTML({
+  title: "Test labeled devices or speakers aren't exposed in enumerateDevices() after fake getUserMedia()",
+  bug: "1743524"
+});
+
+runTest(async () => {
+  await pushPrefs(
+    ["media.setsinkid.enabled", true],
+    // This test uses real devices because fake devices are not grouped with
+    // audiooutput devices.
+    ["media.navigator.streams.fake", false]);
+  const devices = navigator.mediaDevices;
+  {
+    // `fake:true` means that getUserMedia() resolves without any permission
+    // check, and so this should not be sufficient to expose real device info.
+    const stream = await devices.getUserMedia({ audio: true, fake: true });
+    stream.getTracks()[0].stop();
+    const list = await devices.enumerateDevices();
+    const labeledDevices = list.filter(({label}) => label != "");
+    is(labeledDevices.length, 0, "must be zero labeled devices after fake gUM");
+    const outputDevices = list.filter(({kind}) => kind == "audiooutput");
+    is(outputDevices.length, 0, "must be zero output devices after fake gUM");
+  }
+  {
+    // Check without `fake:true` to verify assumptions about existing devices.
+    let stream;
+    try {
+      stream = await devices.getUserMedia({ audio: true });
+      stream.getTracks()[0].stop();
+    } catch (e) {
+      if (e.name == "NotFoundError" &&
+          navigator.userAgent.includes("Mac OS X")) {
+        todo(false, "Expecting no real audioinput device on Mac test machines");
+        return;
+      }
+      throw e;
+    }
+    {
+      const list = await devices.enumerateDevices();
+      const audioDevices = list.filter(({kind}) => kind.includes("audio"));
+      ok(audioDevices.length, "have audio devices after real gUM");
+      const unlabeledAudioDevices = audioDevices.filter(({label}) => !label);
+      is(unlabeledAudioDevices.length, 0,
+         "must be zero unlabeled audio devices after real gUM");
+
+      const outputDevices = list.filter(({kind}) => kind == "audiooutput");
+      isnot(outputDevices.length, 0, "have output devices after real gUM");
+    }
+  }
+});
+  </script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe.html b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe.html
new file mode 100644
index 0000000000..beea3a4f97
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe.html
@@ -0,0 +1,28 @@
+<!DOCTYPE HTML>
+<html>
+<body>
+<pre id="test">
+<script type="application/javascript">
+/**
+  Runs inside iframe in test_enumerateDevices.html.
+*/
+
+const pushPrefs = (...p) => SpecialPowers.pushPrefEnv({set: p});
+const gUM = c => navigator.mediaDevices.getUserMedia(c);
+
+(async () => {
+  await pushPrefs(["media.navigator.streams.fake", true]);
+
+  // Validate enumerated devices after gUM.
+  for (const track of (await gUM({video: true, audio: true})).getTracks()) {
+    track.stop();
+  }
+
+  const devices = await navigator.mediaDevices.enumerateDevices();
+  parent.postMessage(JSON.stringify(devices), "https://example.com:443");
+
+})().catch(e => setTimeout(() => { throw e; }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe_pre_gum.html b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe_pre_gum.html
new file mode 100644
index 0000000000..f2dc2d1f65
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe_pre_gum.html
@@ -0,0 +1,22 @@
+<!DOCTYPE HTML>
+<html>
+<body>
+<pre id="test">
+<script type="application/javascript">
+/**
+  Runs inside iframe in test_enumerateDevices_legacy.html.
+*/
+
+const pushPrefs = (...p) => SpecialPowers.pushPrefEnv({set: p});
+
+(async () => {
+  await pushPrefs(["media.navigator.streams.fake", true]);
+
+  const devices = await navigator.mediaDevices.enumerateDevices();
+  parent.postMessage(JSON.stringify(devices), "https://example.com:443");
+
+})().catch(e => setTimeout(() => { throw e; }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_enumerateDevices_legacy.html b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_legacy.html
new file mode 100644
index 0000000000..c599f2b599
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_legacy.html
@@ -0,0 +1,147 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({ title: "Run enumerateDevices code", bug: "1046245" });
+/**
+  This is a modified copy of test_enumerateDevices.html testing the
+  enumerateDevices() legacy version and deviceId constraint.
+*/
+
+async function mustSucceedWithStream(msg, f) {
+  try {
+    const stream = await f();
+    for (const track of stream.getTracks()) {
+      track.stop();
+    }
+    ok(true, msg + " must succeed");
+  } catch (e) {
+    is(e.name, null, msg + " must succeed: " + e.message);
+  }
+}
+
+async function mustFailWith(msg, reason, constraint, f) {
+  try {
+    await f();
+    ok(false, msg + " must fail");
+  } catch(e) {
+    is(e.name, reason, msg + " must fail: " + e.message);
+    if (constraint) {
+      is(e.constraint, constraint, msg + " must fail w/correct constraint.");
+    }
+  }
+}
+
+const gUM = c => navigator.mediaDevices.getUserMedia(c);
+
+const kinds = ["videoinput", "audioinput", "audiooutput"];
+
+function validateDevice({kind, label, deviceId, groupId}) {
+  ok(kinds.includes(kind), "Known device kind");
+  is(deviceId.length, 44, "deviceId length id as expected for Firefox");
+  ok(label.length !== undefined, "Device label: " + label);
+  isnot(groupId, "", "groupId must be present.");
+}
+
+runTest(async () => {
+  await pushPrefs(["media.navigator.streams.fake", true],
+                  ["media.devices.enumerate.legacy.enabled", true]);
+
+  // Validate enumerated devices before gUM (legacy).
+
+  let devices = await navigator.mediaDevices.enumerateDevices();
+  ok(devices.length, "At least one device found");
+  const jsoned = JSON.parse(JSON.stringify(devices));
+  is(jsoned[0].kind, devices[0].kind, "kind survived serializer");
+  is(jsoned[0].deviceId, devices[0].deviceId, "deviceId survived serializer");
+  for (const device of devices) {
+    validateDevice(device);
+    if (device.kind == "audiooutput") continue;
+    is(device.label, "", "Device label is empty");
+    // Test deviceId constraint
+    let deviceId = device.deviceId;
+    let constraints = (device.kind == "videoinput") ? { video: { deviceId } }
+                                                    : { audio: { deviceId } };
+    let namedDevices;
+    for (const track of (await gUM(constraints)).getTracks()) {
+      is(typeof(track.label), "string", "Track label is a string");
+      isnot(track.label.length, 0, "Track label is not empty");
+      if (!namedDevices) {
+        namedDevices = await navigator.mediaDevices.enumerateDevices();
+      }
+      const namedDevice = namedDevices.find(d => d.deviceId == device.deviceId);
+      is(track.label, namedDevice.label, "Track label is the device label");
+      track.stop();
+    }
+  }
+
+  const unknownId = "unknown9qHr8B0JIbcHlbl9xR+jMbZZ8WyoPfpCXPfc=";
+
+  // Check deviceId failure paths for video.
+
+  await mustSucceedWithStream("unknown plain deviceId on video",
+                              () => gUM({ video: { deviceId: unknownId } }));
+  await mustSucceedWithStream("unknown plain deviceId on audio",
+                              () => gUM({ audio: { deviceId: unknownId } }));
+  await mustFailWith("unknown exact deviceId on video",
+                     "OverconstrainedError", "deviceId",
+                     () => gUM({ video: { deviceId: { exact: unknownId } } }));
+  await mustFailWith("unknown exact deviceId on audio",
+                     "OverconstrainedError", "deviceId",
+                     () => gUM({ audio: { deviceId: { exact: unknownId } } }));
+
+  // Check that deviceIds are stable for same origin and differ across origins.
+
+  const path = "/tests/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe_pre_gum.html";
+  const origins = ["https://example.com", "https://test1.example.com"];
+  info(window.location);
+
+  const haveDevicesMap = new Promise(resolve => {
+    const map = new Map();
+    window.addEventListener("message", ({origin, data}) => {
+      ok(origins.includes(origin), "Got message from expected origin");
+      map.set(origin, JSON.parse(data));
+      if (map.size < origins.length) return;
+      resolve(map);
+    });
+  });
+
+  await Promise.all(origins.map(origin => {
+    const iframe = document.createElement("iframe");
+    iframe.src = origin + path;
+    iframe.allow = "camera;microphone;speaker-selection";
+    info(iframe.src);
+    document.documentElement.appendChild(iframe);
+    return new Promise(resolve => iframe.onload = resolve);
+  }));
+  let devicesMap = await haveDevicesMap;
+  let [sameOriginDevices, differentOriginDevices] = origins.map(o => devicesMap.get(o));
+
+  is(sameOriginDevices.length, devices.length, "same origin same devices");
+  is(differentOriginDevices.length, devices.length, "cross origin same devices");
+  [...sameOriginDevices, ...differentOriginDevices].forEach(d => validateDevice(d));
+
+  for (const device of sameOriginDevices) {
+    ok(devices.find(d => d.deviceId == device.deviceId),
+       "Same origin deviceId for " + device.label + " must match");
+  }
+  for (const device of differentOriginDevices) {
+    ok(!devices.find(d => d.deviceId == device.deviceId),
+         "Different origin deviceId for " + device.label + " must be different");
+  }
+
+  // Check the special case of no devices found.
+  await pushPrefs(["media.navigator.streams.fake", false],
+                  ["media.audio_loopback_dev", "none"],
+                  ["media.video_loopback_dev", "none"]);
+  devices = await navigator.mediaDevices.enumerateDevices();
+  is(devices.length, 0, "No devices");
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_enumerateDevices_navigation.html b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_navigation.html
new file mode 100644
index 0000000000..bf7650223f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_navigation.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<iframe id="iframe" srcdoc="<script>
+  window.enumerateDevices = () =>
+    navigator.mediaDevices.enumerateDevices();
+  </script>"
+    width="100%" height="50%" frameborder="1">
+</iframe>
+<pre id="test">
+<script type="application/javascript">
+createHTML({ title: "Suspend enumerateDevices code ", bug: "1479840" });
+/**
+  This test covers the case that the enumerateDevices method is suspended by
+  navigating away the current window. In order to implement that the enumeration
+  is executed in an iframe which is cleared before the enumeration has been resolved
+*/
+
+runTest(async () => {
+  // Run enumerate devices and mesure the time it will take.
+  const start = new Date().getTime();
+  try {
+    await iframe.contentWindow.enumerateDevices();
+  } catch (e) {
+    info("Failed to enumerate devices, error: " + e);
+  }
+  const elapsed = new Date().getTime() - start;
+
+  // Run again and navigate away. Expected to remain pending.
+  let p = iframe.contentWindow.enumerateDevices()
+  p.then( devices => {
+    ok(false, "Enumerate devices promise resolved unexpectedly, found " + devices.length + " devices.");
+  })
+  .catch ( error => {
+    ok(false, "Enumerate devices promise rejected unexpectedly: " + error);
+  });
+  iframe.srcdoc = "";
+
+  // Wait enough time.
+  try {
+    await timeout(p, 5 * elapsed, "timeout");
+    ok(false, "Enumerate devices promise resolved unexpectedly");
+  } catch (e) {
+    is(e.message, "timeout", "We should time out without enumerateDevices rejecting");
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_fingerprinting_resistance.html b/dom/media/webrtc/tests/mochitests/test_fingerprinting_resistance.html
new file mode 100644
index 0000000000..7e9cd5a219
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_fingerprinting_resistance.html
@@ -0,0 +1,112 @@
+<!DOCTYPE html>
+<html>
+<head>
+<meta charset="utf-8">
+<script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<script>
+/* global SimpleTest SpecialPowers */
+
+async function testEnumerateDevices(expectDevices) {
+  let devices = await navigator.mediaDevices.enumerateDevices();
+  if (!expectDevices) {
+    SimpleTest.is(devices.length, 0, "testEnumerateDevices: No devices");
+    return;
+  }
+  let cams = devices.filter((device) => device.kind == "videoinput");
+  let mics = devices.filter((device) => device.kind == "audioinput");
+  SimpleTest.ok((cams.length == 1) && (mics.length == 1),
+                "testEnumerateDevices: a microphone and a camera");
+}
+
+async function testGetUserMedia(expectDevices) {
+  const constraints = [
+    {audio: true},
+    {video: true},
+    {audio: true, video: true},
+    {video: {width: {min: 1e9}}}, // impossible
+    {audio: {channelCount: {exact: 1e3}}}, // impossible
+  ];
+  for (let constraint of constraints) {
+    let message = "getUserMedia(" + JSON.stringify(constraint) + ")";
+    try {
+      let stream = await navigator.mediaDevices.getUserMedia(constraint);
+      SimpleTest.ok(expectDevices, message + " resolved");
+      if (!expectDevices) {
+        continue;
+      }
+
+      // We only do testGetUserMedia(true) when privacy.resistFingerprinting
+      // is true, test if MediaStreamTrack.label is spoofed.
+      for (let track of stream.getTracks()) {
+        switch (track.kind) {
+          case "audio":
+            SimpleTest.is(track.label, "Internal Microphone", "AudioStreamTrack.label");
+            break;
+          case "video":
+            SimpleTest.is(track.label, "Internal Camera", "VideoStreamTrack.label");
+            break;
+          default:
+            SimpleTest.ok(false, "Unknown kind: " + track.kind);
+            break;
+        }
+        track.stop();
+      }
+    } catch (e) {
+      if (!expectDevices) {
+        SimpleTest.is(e.name, "NotAllowedError", message + " throws NotAllowedError");
+      } else {
+        SimpleTest.ok(false, message + " failed: " + e);
+      }
+    }
+  }
+}
+
+async function testDevices() {
+  await SpecialPowers.pushPrefEnv({
+    set: [
+      ["privacy.resistFingerprinting", true],
+      ["media.navigator.streams.fake", true]
+    ]
+  });
+  await testEnumerateDevices(true); // should list a microphone and a camera
+  await testGetUserMedia(true); // should get audio and video streams
+}
+
+async function testNoDevices() {
+  await SpecialPowers.pushPrefEnv({
+    set: [
+      ["privacy.resistFingerprinting", false],
+      ["media.navigator.permission.device", false],
+      ["media.navigator.streams.fake", false],
+      ["media.audio_loopback_dev", "foo"],
+      ["media.video_loopback_dev", "bar"]
+    ]
+  });
+  await testEnumerateDevices(false); // should list nothing
+  await SpecialPowers.pushPrefEnv({
+    set: [
+      ["privacy.resistFingerprinting", true]
+    ]
+  });
+  await testEnumerateDevices(true); // should list a microphone and a camera
+  await testGetUserMedia(false); // should reject with NotAllowedError
+}
+
+createHTML({
+  title: "Neutralize the threat of fingerprinting of media devices API when 'privacy.resistFingerprinting' is true",
+  bug: "1372073"
+});
+
+runTest(async () => {
+  // Make sure enumerateDevices and getUserMedia work when
+  // privacy.resistFingerprinting is true.
+  await testDevices();
+
+  // Test that absence of devices can't be detected.
+  await testNoDevices();
+});
+</script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_forceSampleRate.html b/dom/media/webrtc/tests/mochitests/test_forceSampleRate.html
new file mode 100644
index 0000000000..c5a9820aaa
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_forceSampleRate.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the pref media.cubeb.force_sample_rate</title>
+  <script type="text/javascript" src="/MochiKit/MochiKit.js"></script>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script class="testbody" type="text/javascript">
+const WEIRD_SAMPLE_RATE = 44101;
+
+SimpleTest.waitForExplicitFinish();
+SpecialPowers.pushPrefEnv({"set": [
+  ["media.cubeb.force_sample_rate", WEIRD_SAMPLE_RATE]
+]}).then(function() {
+  var ac = new AudioContext();
+  is(ac.sampleRate, WEIRD_SAMPLE_RATE, "Forced sample-rate set successfully.");
+  SimpleTest.finish();
+});
+</script>
+</pre>
+</body>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_GC_MediaStream.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_GC_MediaStream.html
new file mode 100644
index 0000000000..5aa0e64947
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_GC_MediaStream.html
@@ -0,0 +1,59 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  "use strict";
+
+  createHTML({
+    title: "MediaStreams can be garbage collected",
+    bug: "1407542"
+  });
+
+  let SpecialStream = SpecialPowers.wrap(MediaStream);
+
+  async function testGC(stream, numCopies, copy) {
+    let startStreams = await SpecialStream.countUnderlyingStreams();
+
+    let copies = new Array(numCopies).fill(0).map(() => copy(stream));
+    ok(await SpecialStream.countUnderlyingStreams() > startStreams,
+        "MediaStreamTrack constructor creates more underlying streams");
+
+    copies = [];
+    await new Promise(r => SpecialPowers.exactGC(r));
+    is(await SpecialStream.countUnderlyingStreams(), startStreams,
+       "MediaStreamTracks should have been collected");
+  }
+
+  runTest(async () => {
+    // We do not need LoopbackTone because it is not used
+    // and creates extra streams that affect the result
+    DISABLE_LOOPBACK_TONE = true;
+
+    let gUMStream = await getUserMedia({video: true});
+    info("Testing GC of track-array constructor with cloned tracks");
+    await testGC(gUMStream, 10, s => new MediaStream(s.getTracks().map(t => t.clone())));
+
+    info("Testing GC of empty constructor plus addTrack with cloned tracks");
+    await testGC(gUMStream, 10, s => {
+      let s2 = new MediaStream();
+      s.getTracks().forEach(t => s2.addTrack(t.clone()));
+      return s2;
+    });
+
+    info("Testing GC of cloned stream");
+    await testGC(gUMStream, 10, s => s.clone());
+
+    info("Testing GC of gUM stream");
+    gUMStream = null;
+    await new Promise(r => SpecialPowers.exactGC(r));
+    is(await SpecialStream.countUnderlyingStreams(), 0,
+       "Original gUM stream should be collectable");
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_active_autoplay.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_active_autoplay.html
new file mode 100644
index 0000000000..c1a39cdd4c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_active_autoplay.html
@@ -0,0 +1,61 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<video id="testAutoplay" autoplay></video>
+<script type="application/javascript">
+"use strict";
+
+const video = document.getElementById("testAutoplay");
+var stream;
+var otherVideoTrack;
+var otherAudioTrack;
+
+createHTML({
+  title: "MediaStream can be autoplayed in media element after going inactive and then active",
+  bug: "1208316"
+});
+
+runTest(() => getUserMedia({audio: true, video: true}).then(s => {
+  stream = s;
+  otherVideoTrack = stream.getVideoTracks()[0].clone();
+  otherAudioTrack = stream.getAudioTracks()[0].clone();
+
+  video.srcObject = stream;
+  return haveEvent(video, "playing", wait(5000, new Error("Timeout")));
+})
+.then(() => {
+  ok(!video.ended, "Video element should be playing after adding a gUM stream");
+  stream.getTracks().forEach(t => t.stop());
+  return haveEvent(video, "ended", wait(5000, new Error("Timeout")));
+})
+.then(() => {
+  ok(video.ended, "Video element should be ended");
+  stream.addTrack(otherVideoTrack);
+  return haveEvent(video, "playing", wait(5000, new Error("Timeout")));
+})
+.then(() => {
+  ok(!video.ended, "Video element should be playing after adding a video track");
+  stream.getTracks().forEach(t => t.stop());
+  return haveEvent(video, "ended", wait(5000, new Error("Timeout")));
+})
+.then(() => {
+  ok(video.ended, "Video element should be ended");
+  stream.addTrack(otherAudioTrack);
+  return haveEvent(video, "playing", wait(5000, new Error("Timeout")));
+})
+.then(() => {
+  ok(!video.ended, "Video element should be playing after adding a audio track");
+  stream.getTracks().forEach(t => t.stop());
+  return haveEvent(video, "ended", wait(5000, new Error("Timeout")));
+})
+.then(() => {
+  ok(video.ended, "Video element should be ended");
+}));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_addTrackRemoveTrack.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_addTrackRemoveTrack.html
new file mode 100644
index 0000000000..27dad2519f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_addTrackRemoveTrack.html
@@ -0,0 +1,169 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  "use strict";
+
+  createHTML({
+    title: "MediaStream's addTrack() and removeTrack() with getUserMedia streams Test",
+    bug: "1103188"
+  });
+
+  runTest(() => Promise.resolve()
+    .then(() => getUserMedia({audio: true})).then(stream =>
+      getUserMedia({video: true}).then(otherStream => {
+        info("Test addTrack()ing a video track to an audio-only gUM stream");
+        var track = stream.getTracks()[0];
+        var otherTrack = otherStream.getTracks()[0];
+
+        stream.addTrack(track);
+        checkMediaStreamContains(stream, [track], "Re-added audio");
+
+        stream.addTrack(otherTrack);
+        checkMediaStreamContains(stream, [track, otherTrack], "Added video");
+
+        var testElem = createMediaElement('video', 'testAddTrackAudioVideo');
+        var playback = new MediaStreamPlayback(testElem, stream);
+        return playback.playMedia(false);
+    }))
+    .then(() => getUserMedia({video: true})).then(stream =>
+      getUserMedia({video: true}).then(otherStream => {
+        info("Test addTrack()ing a video track to a video-only gUM stream");
+        var track = stream.getTracks()[0];
+        var otherTrack = otherStream.getTracks()[0];
+
+        stream.addTrack(track);
+        checkMediaStreamContains(stream, [track], "Re-added video");
+
+        stream.addTrack(otherTrack);
+        checkMediaStreamContains(stream, [track, otherTrack], "Added video");
+
+        var test = createMediaElement('video', 'testAddTrackDoubleVideo');
+        var playback = new MediaStreamPlayback(test, stream);
+        return playback.playMedia(false);
+    }))
+    .then(() => getUserMedia({video: true})).then(stream =>
+      getUserMedia({video: true}).then(otherStream => {
+        info("Test removeTrack() existing and added video tracks from a video-only gUM stream");
+        var track = stream.getTracks()[0];
+        var otherTrack = otherStream.getTracks()[0];
+
+        stream.removeTrack(otherTrack);
+        checkMediaStreamContains(stream, [track], "Removed non-existing video");
+
+        stream.addTrack(otherTrack);
+        checkMediaStreamContains(stream, [track, otherTrack], "Added video");
+
+        stream.removeTrack(otherTrack);
+        checkMediaStreamContains(stream, [track], "Removed added video");
+
+        stream.removeTrack(otherTrack);
+        checkMediaStreamContains(stream, [track], "Re-removed added video");
+
+        stream.removeTrack(track);
+        checkMediaStreamContains(stream, [], "Removed original video");
+
+        var elem = createMediaElement('video', 'testRemoveAllVideo');
+        var loadeddata = false;
+        elem.onloadeddata = () => { loadeddata = true; elem.onloadeddata = null; };
+        elem.srcObject = stream;
+        elem.play();
+        return wait(500).then(() => {
+          ok(!loadeddata, "Stream without tracks shall not raise 'loadeddata' on media element");
+          elem.pause();
+          elem.srcObject = null;
+        })
+        .then(() => {
+          stream.addTrack(track);
+          checkMediaStreamContains(stream, [track], "Re-added added-then-removed track");
+          var playback = new MediaStreamPlayback(elem, stream);
+          return playback.playMedia(false);
+        })
+        .then(() => otherTrack.stop());
+    }))
+    .then(() => getUserMedia({ audio: true })).then(audioStream =>
+      getUserMedia({ video: true }).then(videoStream => {
+        info("Test adding track and removing the original");
+        var audioTrack = audioStream.getTracks()[0];
+        var videoTrack = videoStream.getTracks()[0];
+        videoStream.removeTrack(videoTrack);
+        audioStream.addTrack(videoTrack);
+
+        checkMediaStreamContains(videoStream, [], "1, Removed original track");
+        checkMediaStreamContains(audioStream, [audioTrack, videoTrack],
+                                 "2, Added external track");
+
+        var elem = createMediaElement('video', 'testAddRemoveOriginalTrackVideo');
+        var playback = new MediaStreamPlayback(elem, audioStream);
+        return playback.playMedia(false);
+      }))
+    .then(() => getUserMedia({ audio: true, video: true })).then(stream => {
+      info("Test removing stopped tracks");
+      stream.getTracks().forEach(t => {
+        t.stop();
+        stream.removeTrack(t);
+      });
+      checkMediaStreamContains(stream, [], "Removed stopped tracks");
+    })
+    .then(() => {
+      var ac = new AudioContext();
+
+      var osc1k = createOscillatorStream(ac, 1000);
+      var audioTrack1k = osc1k.getTracks()[0];
+
+      var osc5k = createOscillatorStream(ac, 5000);
+      var audioTrack5k = osc5k.getTracks()[0];
+
+      var osc10k = createOscillatorStream(ac, 10000);
+      var audioTrack10k = osc10k.getTracks()[0];
+
+      var stream = osc1k;
+      return Promise.resolve().then(() => {
+        info("Analysing audio output with original 1k track");
+        var analyser = new AudioStreamAnalyser(ac, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(50)] < 50 &&
+                 array[analyser.binIndexForFrequency(1000)] > 200 &&
+                 array[analyser.binIndexForFrequency(5000)] < 50 &&
+                 array[analyser.binIndexForFrequency(10000)] < 50);
+      }).then(() => {
+        info("Analysing audio output with removed original 1k track and added 5k track");
+        stream.removeTrack(audioTrack1k);
+        stream.addTrack(audioTrack5k);
+        var analyser = new AudioStreamAnalyser(ac, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(1000)] < 50 &&
+                 array[analyser.binIndexForFrequency(5000)] > 200 &&
+                 array[analyser.binIndexForFrequency(10000)] < 50);
+      }).then(() => {
+        info("Analysing audio output with removed 5k track and added 10k track");
+        stream.removeTrack(audioTrack5k);
+        stream.addTrack(audioTrack10k);
+        var analyser = new AudioStreamAnalyser(ac, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(1000)] < 50 &&
+                 array[analyser.binIndexForFrequency(5000)] < 50 &&
+                 array[analyser.binIndexForFrequency(10000)] > 200);
+      }).then(() => {
+        info("Analysing audio output with re-added 1k, 5k and added 10k tracks");
+        stream.addTrack(audioTrack1k);
+        stream.addTrack(audioTrack5k);
+        var analyser = new AudioStreamAnalyser(ac, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(50)] < 50 &&
+                 array[analyser.binIndexForFrequency(1000)] > 200 &&
+                 array[analyser.binIndexForFrequency(2500)] < 50 &&
+                 array[analyser.binIndexForFrequency(5000)] > 200 &&
+                 array[analyser.binIndexForFrequency(7500)] < 50 &&
+                 array[analyser.binIndexForFrequency(10000)] > 200 &&
+                 array[analyser.binIndexForFrequency(11000)] < 50);
+      });
+    }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_addtrack_removetrack_events.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_addtrack_removetrack_events.html
new file mode 100644
index 0000000000..833653ebb2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_addtrack_removetrack_events.html
@@ -0,0 +1,110 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "MediaStream's 'addtrack' and 'removetrack' events shouldn't fire on manual operations",
+  bug: "1208328"
+});
+
+var spinEventLoop = () => new Promise(r => setTimeout(r, 0));
+
+var stream;
+var clone;
+var newStream;
+var tracks = [];
+
+var addTrack = track => {
+  info("Adding track " + track.id);
+  stream.addTrack(track);
+};
+var removeTrack = track => {
+  info("Removing track " + track.id);
+  stream.removeTrack(track);
+};
+var stopTrack = track => {
+  if (track.readyState == "live") {
+    info("Stopping track " + track.id);
+  }
+  track.stop();
+};
+
+runTest(() => getUserMedia({audio: true, video: true})
+  .then(s => {
+    stream = s;
+    clone = s.clone();
+    stream.addEventListener("addtrack", function onAddtrack(event) {
+      ok(false, "addtrack fired unexpectedly for track " + event.track.id);
+    });
+    stream.addEventListener("removetrack", function onRemovetrack(event) {
+      ok(false, "removetrack fired unexpectedly for track " + event.track.id);
+    });
+
+    return getUserMedia({audio: true, video: true});
+  })
+  .then(s => {
+    newStream = s;
+    info("Stopping an original track");
+    stopTrack(stream.getTracks()[0]);
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Removing original tracks");
+    stream.getTracks().forEach(t => (stream.removeTrack(t), tracks.push(t)));
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Adding other gUM tracks");
+    newStream.getTracks().forEach(t => addTrack(t))
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Adding cloned tracks");
+    let clone = stream.clone();
+    clone.getTracks().forEach(t => addTrack(t));
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Removing a clone");
+    removeTrack(clone.getTracks()[0]);
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Stopping clones");
+    clone.getTracks().forEach(t => stopTrack(t));
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Stopping originals");
+    stream.getTracks().forEach(t => stopTrack(t));
+    tracks.forEach(t => stopTrack(t));
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Removing remaining tracks");
+    stream.getTracks().forEach(t => removeTrack(t));
+
+    return spinEventLoop();
+  })
+  .then(() => {
+      // Test MediaStreamTrackEvent required args here.
+      mustThrowWith("MediaStreamTrackEvent without required args",
+                    "TypeError", () => new MediaStreamTrackEvent("addtrack", {}));
+  }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioCapture.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioCapture.html
new file mode 100644
index 0000000000..2cc649a321
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioCapture.html
@@ -0,0 +1,104 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioCapture </title>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script>
+
+(async () => {
+  // Get an opus file containing a sine wave at maximum amplitude, of duration
+  // `lengthSeconds`, and of frequency `frequency`.
+  async function getSineWaveFile(frequency, lengthSeconds) {
+    const off = new OfflineAudioContext(1, lengthSeconds * 48000, 48000);
+    const osc = off.createOscillator();
+    const rec = new MediaRecorder(off.destination,
+                                  {mimeType: "audio/ogg; codecs=opus"});
+    osc.frequency.value = frequency;
+    osc.connect(off.destination);
+    osc.start();
+    rec.start();
+    off.startRendering();
+    const {data} = await new Promise(r => rec.ondataavailable = r);
+    return data;
+  }
+
+  await createHTML({
+    bug: "1156472",
+    title: "Test AudioCapture with regular HTMLMediaElement, AudioContext, " +
+           "and HTMLMediaElement playing a MediaStream",
+    visible: true
+  });
+
+  await runTestWhenReady(async () => {
+    /**
+     * Get two HTMLMediaElements:
+     * - One playing a sine tone from a blob (of an opus file created on the fly)
+     * - One being the output for an AudioContext's OscillatorNode, connected to
+     *   a MediaSourceDestinationNode.
+     *
+     * Also, use the AudioContext playing through its AudioDestinationNode another
+     * tone, using another OscillatorNode.
+     *
+     * Capture the output of the document, feed that back into the AudioContext,
+     * with an AnalyserNode, and check the frequency content to make sure we
+     * have recorded the three sources.
+     *
+     * The three sine tones have frequencies far apart from each other, so that we
+     * can check that the spectrum of the capture stream contains three
+     * components with a high magnitude.
+     */
+    const wavtone = createMediaElement("audio", "WaveTone");
+    const acTone = createMediaElement("audio", "audioContextTone");
+    const ac = new AudioContext();
+
+    const oscThroughMediaElement = ac.createOscillator();
+    oscThroughMediaElement.frequency.value = 1000;
+    const oscThroughAudioDestinationNode = ac.createOscillator();
+    oscThroughAudioDestinationNode.frequency.value = 5000;
+    const msDest = ac.createMediaStreamDestination();
+
+    oscThroughMediaElement.connect(msDest);
+    oscThroughAudioDestinationNode.connect(ac.destination);
+
+    acTone.srcObject = msDest.stream;
+
+    const blob = await getSineWaveFile(10000, 10);
+    wavtone.src = URL.createObjectURL(blob);
+    oscThroughMediaElement.start();
+    oscThroughAudioDestinationNode.start();
+    wavtone.loop = true;
+    wavtone.play();
+    acTone.play();
+
+    const constraints = {audio: {mediaSource: "audioCapture"}};
+
+    const stream = await getUserMedia(constraints);
+    try {
+      const analyser = new AudioStreamAnalyser(ac, stream);
+      analyser.enableDebugCanvas();
+      await analyser.waitForAnalysisSuccess(array => {
+        // We want to find three frequency components here, around 1000, 5000
+        // and 10000Hz. Frequency are logarithmic. Also make sure we have low
+        // energy in between, not just a flat white noise.
+        return (array[analyser.binIndexForFrequency(50)]    < 50 &&
+                array[analyser.binIndexForFrequency(1000)]  > 200 &&
+                array[analyser.binIndexForFrequency(2500)]  < 50 &&
+                array[analyser.binIndexForFrequency(5000)]  > 200 &&
+                array[analyser.binIndexForFrequency(7500)]  < 50 &&
+                array[analyser.binIndexForFrequency(10000)] > 200);
+      });
+    } finally {
+      for (let t of stream.getTracks()) {
+        t.stop();
+      }
+      ac.close();
+    }
+  });
+})();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints.html
new file mode 100644
index 0000000000..162e83063a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints.html
@@ -0,0 +1,93 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "Test that microphone getSettings report correct settings after applyConstraints",
+  bug: "1447982",
+});
+
+function testTrackAgainstAudioConstraints(track, audioConstraints) {
+  let constraints = track.getConstraints();
+  is(constraints.autoGainControl, audioConstraints.autoGainControl,
+     "Should report correct autoGainControl constraint");
+  is(constraints.echoCancellation, audioConstraints.echoCancellation,
+     "Should report correct echoCancellation constraint");
+  is(constraints.noiseSuppression, audioConstraints.noiseSuppression,
+     "Should report correct noiseSuppression constraint");
+
+  let settings = track.getSettings();
+  is(settings.autoGainControl, audioConstraints.autoGainControl,
+     "Should report correct autoGainControl setting");
+  is(settings.echoCancellation, audioConstraints.echoCancellation,
+     "Should report correct echoCancellation setting");
+  is(settings.noiseSuppression, audioConstraints.noiseSuppression,
+     "Should report correct noiseSuppression setting");
+}
+
+async function testAudioConstraints(track, audioConstraints) {
+  // We applyConstraints() first and do a fresh gUM later, to avoid
+  // testing multiple concurrent captures at different settings.
+
+  info(`Testing applying constraints ${JSON.stringify(audioConstraints)} ` +
+       `to track with settings ${JSON.stringify(track.getSettings())}`);
+  await track.applyConstraints(audioConstraints);
+  testTrackAgainstAudioConstraints(track, audioConstraints);
+
+  info("Testing fresh gUM request with audio constraints " +
+       JSON.stringify(audioConstraints));
+  let stream = await getUserMedia({audio: audioConstraints});
+  testTrackAgainstAudioConstraints(stream.getTracks()[0], audioConstraints);
+  stream.getTracks().forEach(t => t.stop());
+}
+
+runTest(async () => {
+  let audioDevice = SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (!audioDevice) {
+    ok(false, "No device set by framework. Try --use-test-media-devices");
+    return;
+  }
+
+  let supportedConstraints = navigator.mediaDevices.getSupportedConstraints();
+  is(supportedConstraints.autoGainControl, true,
+     "autoGainControl constraint should be supported");
+  is(supportedConstraints.echoCancellation, true,
+     "echoCancellation constraint should be supported");
+  is(supportedConstraints.noiseSuppression, true,
+     "noiseSuppression constraint should be supported");
+
+  let egn = (e, g, n) => ({
+    echoCancellation: e,
+    autoGainControl: g,
+    noiseSuppression: n
+  });
+
+  let stream = await getUserMedia({
+    audio: egn(true, true, true),
+  });
+  let track = stream.getTracks()[0];
+  let audioConstraintsToTest = [
+    egn(false, true,  true),
+    egn(true,  false, true),
+    egn(true,  true,  false),
+    egn(false, false, true),
+    egn(false, true,  false),
+    egn(true,  false, false),
+    egn(false, false, false),
+    egn(true,  true,  true),
+  ];
+  for (let audioConstraints of audioConstraintsToTest) {
+    await testAudioConstraints(track, audioConstraints);
+  }
+  track.stop();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints_concurrentIframes.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints_concurrentIframes.html
new file mode 100644
index 0000000000..d07dbc41f1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints_concurrentIframes.html
@@ -0,0 +1,157 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "getUserMedia in multiple iframes with different constraints",
+  bug: "1404977"
+});
+/**
+  * Verify that we can successfully call getUserMedia for the same device in
+  * multiple iframes concurrently. This is checked by creating a number of
+  * iframes and performing a separate getUserMedia call in each. We verify the
+  * stream returned by that call has the same constraints as requested both
+  * immediately after the call and after all gUM calls have been made. The test
+  * then verifies the streams can be played.
+  */
+runTest(async function() {
+  // Compare constraints and return a string with the differences in
+  // echoCancellation, autoGainControl, and noiseSuppression. The string
+  // will be empty if there are no differences.
+  function getConstraintDifferenceString(constraints, otherConstraints) {
+    let diffString = "";
+    if (constraints.echoCancellation != otherConstraints.echoCancellation) {
+      diffString += "echoCancellation different: " +
+                    `${constraints.echoCancellation} != ${otherConstraints.echoCancellation}, `;
+    }
+    if (constraints.autoGainControl != otherConstraints.autoGainControl) {
+      diffString += "autoGainControl different: " +
+                    `${constraints.autoGainControl} != ${otherConstraints.autoGainControl}, `;
+    }
+    if (constraints.noiseSuppression != otherConstraints.noiseSuppression) {
+      diffString += "noiseSuppression different: " +
+                    `${constraints.noiseSuppression} != ${otherConstraints.noiseSuppression}, `;
+    }
+    // Replace trailing comma and space if any
+    return diffString.replace(/, $/, "");
+  }
+
+  // We need a real device to get a MediaEngine supporting constraints
+  let audioDevice = SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (!audioDevice) {
+    todo(false, "No device set by framework. Try --use-test-media-devices");
+    return;
+  }
+
+  let egn = (e, g, n) => ({
+    echoCancellation: e,
+    autoGainControl: g,
+    noiseSuppression: n
+  });
+
+  let allConstraintCombinations = [
+    egn(false, false, false),
+    egn(true,  false, false),
+    egn(false, true,  false),
+    egn(false, false, true),
+    egn(true,  true,  false),
+    egn(true,  false, true),
+    egn(false, true,  true),
+    egn(true,  true,  true),
+  ];
+
+  // TODO: We would like to be able to perform an arbitrary number of gUM calls
+  // at once, but issues with pulse and audio IPC mean on some systems we're
+  // limited to as few as 2 concurrent calls. To avoid issues we chunk test runs
+  // to only two calls at a time. The while, splice and GC lines can be removed,
+  // the extra scope removed and allConstraintCombinations can be renamed to
+  // constraintCombinations once this issue is resolved. See bug 1480489.
+  while (allConstraintCombinations.length) {
+    {
+      let constraintCombinations = allConstraintCombinations.splice(0, 2);
+      // Array to store objects that associate information used in our test such as
+      // constraints, iframes, gum streams, and various promises.
+      let testCases = [];
+
+      for (let constraints of constraintCombinations) {
+        let testCase = {requestedConstraints: constraints};
+        // Provide an id for logging, labeling related elements.
+        testCase.id = `testCase.` +
+                      `e=${constraints.echoCancellation}.` +
+                      `g=${constraints.noiseSuppression}.` +
+                      `n=${constraints.noiseSuppression}`;
+        testCases.push(testCase);
+        testCase.iframe = document.createElement("iframe");
+        testCase.iframeLoadedPromise = new Promise((resolve, reject) => {
+          testCase.iframe.onload = () => { resolve(); };
+        });
+        document.body.appendChild(testCase.iframe);
+      }
+      is(testCases.length,
+        constraintCombinations.length,
+        "Should have created a testcase for each constraint");
+
+      // Wait for all iframes to be loaded
+      await Promise.all(testCases.map(tc => tc.iframeLoadedPromise));
+
+      // Start a tone at our top level page so the gUM calls will record something
+      // should we wish to verify their recording in future.
+      let tone = new LoopbackTone(new AudioContext, TEST_AUDIO_FREQ);
+      tone.start();
+
+      // One by one see if we can grab a gUM stream per iframe
+      for (let testCase of testCases) {
+        // Use normal gUM rather than our test helper as the test harness was
+        // not made to be used inside iframes.
+        testCase.gumStream =
+          await testCase.iframe.contentWindow.navigator.mediaDevices.getUserMedia({audio: testCase.requestedConstraints})
+          .catch(e => Promise.reject(`getUserMedia calls should not fail! Failed at ${testCase.id} with: ${e}!`));
+        let differenceString = getConstraintDifferenceString(
+          testCase.requestedConstraints,
+          testCase.gumStream.getAudioTracks()[0].getSettings());
+        ok(!differenceString,
+          `gUM stream for ${testCase.id} should have the same constraints as were ` +
+          `requested from gUM. Differences: ${differenceString}`);
+      }
+
+      // Once all streams are collected, make sure the constraints haven't been
+      // mutated by another gUM call.
+      for (let testCase of testCases) {
+        let differenceString = getConstraintDifferenceString(
+          testCase.requestedConstraints,
+          testCase.gumStream.getAudioTracks()[0].getSettings());
+        ok(!differenceString,
+          `gUM stream for ${testCase.id} should not have had constraints altered after ` +
+          `all gUM calls are done. Differences: ${differenceString}`);
+      }
+
+      // We do not currently have tests to verify the behaviour of the different
+      // constraints. Once we do we should do further verification here. See
+      // bug 1406372, bug 1406376, and bug 1406377.
+
+      for (let testCase of testCases) {
+        let testAudio = createMediaElement("audio", `testAudio.${testCase.id}`);
+        let playback = new MediaStreamPlayback(testAudio, testCase.gumStream);
+        await playback.playMediaWithoutStoppingTracks(false);
+      }
+
+      // Stop the tracks for each stream, we left them running above via
+      // playMediaWithoutStoppingTracks to make sure they can play concurrently.
+      for (let testCase of testCases) {
+        testCase.gumStream.getTracks().map(t => t.stop());
+        document.body.removeChild(testCase.iframe);
+      }
+
+      tone.stop();
+    }
+    await new Promise(r => SpecialPowers.exactGC(r));
+  }
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints_concurrentStreams.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints_concurrentStreams.html
new file mode 100644
index 0000000000..f5b5e784ea
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints_concurrentStreams.html
@@ -0,0 +1,123 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "getUserMedia multiple times, concurrently, and with different constraints",
+  bug: "1404977"
+});
+/**
+  * Verify that we can successfully call getUserMedia multiple times for the
+  * same device, concurrently. This is checked by calling getUserMedia a number
+  * of times with different constraints. We verify that the stream returned by
+  * that call has the same constraints as requested both immediately after the
+  * call and after all gUM calls have been made. The test then verifies the
+  * streams can be played.
+  */
+runTest(async function() {
+  // Compare constraints and return a string with the differences in
+  // echoCancellation, autoGainControl, and noiseSuppression. The string
+  // will be empty if there are no differences.
+  function getConstraintDifferenceString(constraints, otherConstraints) {
+    let diffString = "";
+    if (constraints.echoCancellation != otherConstraints.echoCancellation) {
+      diffString += "echoCancellation different: " +
+                    `${constraints.echoCancellation} != ${otherConstraints.echoCancellation}, `;
+    }
+    if (constraints.autoGainControl != otherConstraints.autoGainControl) {
+      diffString += "autoGainControl different: " +
+                    `${constraints.autoGainControl} != ${otherConstraints.autoGainControl}, `;
+    }
+    if (constraints.noiseSuppression != otherConstraints.noiseSuppression) {
+      diffString += "noiseSuppression different: " +
+                    `${constraints.noiseSuppression} != ${otherConstraints.noiseSuppression}, `;
+    }
+    // Replace trailing comma and space if any
+    return diffString.replace(/, $/, "");
+  }
+
+  // We need a real device to get a MediaEngine supporting constraints
+  let audioDevice = SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (!audioDevice) {
+    todo(false, "No device set by framework. Try --use-test-media-devices");
+    return;
+  }
+
+  let egn = (e, g, n) => ({
+    echoCancellation: e,
+    autoGainControl: g,
+    noiseSuppression: n
+  });
+
+  let constraintCombinations = [
+    egn(false, false, false),
+    egn(true,  false, false),
+    egn(false, true,  false),
+    egn(false, false, true),
+    egn(true,  true,  false),
+    egn(true,  false, true),
+    egn(false, true,  true),
+    egn(true,  true,  true),
+  ];
+
+  // Array to store objects that associate information used in our test such as
+  // constraints, gum streams, and various promises.
+  let testCases = [];
+
+  for (let constraints of constraintCombinations) {
+    let testCase = {requestedConstraints: constraints};
+    // Provide an id for logging, labeling related elements.
+    testCase.id = `testCase.` +
+                  `e=${constraints.echoCancellation}.` +
+                  `g=${constraints.noiseSuppression}.` +
+                  `n=${constraints.noiseSuppression}`;
+    testCases.push(testCase);
+    testCase.gumStream =
+      await getUserMedia({audio: testCase.requestedConstraints})
+      .catch(e => Promise.reject(`getUserMedia calls should not fail! Failed at ${testCase.id} with: ${e}!`));
+    let differenceString = getConstraintDifferenceString(
+      testCase.requestedConstraints,
+      testCase.gumStream.getAudioTracks()[0].getSettings());
+      ok(!differenceString,
+        `gUM stream for ${testCase.id} should have the same constraints as were ` +
+        `requested from gUM. Differences: ${differenceString}`);
+  }
+  is(testCases.length,
+    constraintCombinations.length,
+    "Should have a stream for each constraint");
+
+  // Once all streams are collected, make sure the constraints haven't been
+  // mutated by another gUM call.
+  for (let testCase of testCases) {
+    let differenceString = getConstraintDifferenceString(
+      testCase.requestedConstraints,
+      testCase.gumStream.getAudioTracks()[0].getSettings());
+    ok(!differenceString,
+      `gUM stream for ${testCase.id} should not have had constraints altered after ` +
+      `all gUM calls are done. Differences: ${differenceString}`);
+  }
+
+  // We do not currently have tests to verify the behaviour of the different
+  // constraints. Once we do we should do further verificaiton here. See
+  // bug 1406372, bug 1406376, and bug 1406377.
+
+  for (let testCase of testCases) {
+    let testAudio = createMediaElement("audio", `testAudio.${testCase.id}`);
+    let playback = new MediaStreamPlayback(testAudio, testCase.gumStream);
+    await playback.playMediaWithoutStoppingTracks(false);
+  }
+
+  // Stop the tracks for each stream, we left them running above via
+  // playMediaWithoutStoppingTracks to make sure they can play concurrently.
+  for (let testCase of testCases) {
+    testCase.gumStream.getTracks().map(t => t.stop());
+  }
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicAudio.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicAudio.html
new file mode 100644
index 0000000000..b4775b4244
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicAudio.html
@@ -0,0 +1,27 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({ title: "getUserMedia Basic Audio Test", bug: "781534" });
+  /**
+   * Run a test to verify that we can complete a start and stop media playback
+   * cycle for an audio MediaStream on an audio HTMLMediaElement.
+   */
+  runTest(function () {
+    var testAudio = createMediaElement('audio', 'testAudio');
+    var constraints = {audio: true};
+
+    return getUserMedia(constraints).then(stream => {
+      var playback = new MediaStreamPlayback(testAudio, stream);
+      return playback.playMedia(false);
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicAudio_loopback.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicAudio_loopback.html
new file mode 100644
index 0000000000..10bf669c00
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicAudio_loopback.html
@@ -0,0 +1,99 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+
+<script>
+  createHTML({
+    title: "getUserMedia Basic Audio Test Loopback",
+    bug: "1406350",
+    visible: true
+  });
+
+  /**
+   * Run a test to verify the use of LoopbackTone as audio input.
+   */
+  runTest(async () => {
+    if (!SpecialPowers.getCharPref("media.audio_loopback_dev", "")) {
+      todo(false, "No loopback device set by framework. Try --use-test-media-devices");
+      return;
+    }
+
+    // At this point DefaultLoopbackTone has been instantiated
+    // automatically on frequency TEST_AUDIO_FREQ (440 Hz). Verify
+    // that a tone is detected on that frequency.
+    info("Capturing at default frequency");
+    const stream = await getUserMedia({audio: true});
+
+    try {
+      const audioContext = new AudioContext();
+      const analyser = new AudioStreamAnalyser(audioContext, stream);
+      analyser.enableDebugCanvas();
+      await analyser.waitForAnalysisSuccess(array => {
+        // High energy on 1000 Hz low energy around that
+        const freg_50Hz   = array[analyser.binIndexForFrequency(50)];
+        const freq        = array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)];
+        const freq_2000Hz = array[analyser.binIndexForFrequency(2000)];
+
+        info("Analysing audio frequency - low:target:high = "
+                + freg_50Hz + ':' + freq + ':' + freq_2000Hz);
+        return freg_50Hz < 50 && freq > 200 && freq_2000Hz < 50;
+      });
+
+      // Use the LoopbackTone API to change the frequency of the default tone.
+      // Verify that a tone is detected on the new frequency (800 Hz).
+      info("Change loopback tone frequency");
+      DefaultLoopbackTone.changeFrequency(800);
+      await analyser.waitForAnalysisSuccess(array => {
+        const freg_50Hz   = array[analyser.binIndexForFrequency(50)];
+        const freq        = array[analyser.binIndexForFrequency(800)];
+        const freq_2000Hz = array[analyser.binIndexForFrequency(2000)];
+
+        info("Analysing audio frequency - low:target:high = "
+                + freg_50Hz + ':' + freq + ':' + freq_2000Hz);
+        return freg_50Hz < 50 && freq > 200 && freq_2000Hz < 50;
+      });
+
+      // Create a second tone at a different frequency.
+      // Verify that both tones are detected.
+      info("Multiple loopback tones");
+      DefaultLoopbackTone.changeFrequency(TEST_AUDIO_FREQ);
+      const second_tone = new LoopbackTone(audioContext, 2000);
+      second_tone.start();
+      await analyser.waitForAnalysisSuccess(array => {
+        const freg_50Hz   = array[analyser.binIndexForFrequency(50)];
+        const freq        = array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)];
+        const freq_2000Hz = array[analyser.binIndexForFrequency(2000)];
+        const freq_4000Hz = array[analyser.binIndexForFrequency(4000)];
+
+        info("Analysing audio frequency - low:target1:target2:high = "
+                + freg_50Hz + ':' + freq + ':' + freq_2000Hz + ':' + freq_4000Hz);
+        return freg_50Hz < 50 && freq > 200 && freq_2000Hz > 200 && freq_4000Hz < 50;
+      });
+
+      // Stop all tones and verify that there is no audio on the given frequencies.
+      info("Stop all loopback tones");
+      DefaultLoopbackTone.stop();
+      second_tone.stop()
+      await analyser.waitForAnalysisSuccess(array => {
+        const freg_50Hz   = array[analyser.binIndexForFrequency(50)];
+        const freq        = array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)];
+        const freq_2000Hz = array[analyser.binIndexForFrequency(2000)];
+
+        info("Analysing audio frequency - low:target:high = "
+                + freg_50Hz + ':' + freq + ':' + freq_2000Hz);
+        return freg_50Hz < 50 && freq < 50 && freq_2000Hz < 50;
+      });
+    } finally {
+      for (let t of stream.getTracks()) {
+        t.stop();
+      }
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicScreenshare.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicScreenshare.html
new file mode 100644
index 0000000000..cc73de77da
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicScreenshare.html
@@ -0,0 +1,260 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia Basic Screenshare Test",
+    bug: "1211656",
+    visible: true,
+  });
+
+  const {AppConstants} =
+    SpecialPowers.ChromeUtils.import("resource://gre/modules/AppConstants.jsm");
+
+  // Since the MacOS backend captures in the wrong rgb color profile we need
+  // large thresholds there, and they vary greatly by color. We define
+  // thresholds per platform and color here to still allow the test to run.
+  // Since the colors used (red, green, blue, white) consist only of "pure"
+  // components (0 or 255 for each component), the high thresholds on Mac will
+  // still be able to catch an error where the image is corrupt, or if frames
+  // don't flow.
+  const thresholds = {
+    // macos: captures in the display rgb color profile, which we treat as
+    // sRGB, which is most likely wrong. These thresholds are needed currently
+    // in CI. See bug 1827606.
+    "macosx": { "red": 120, "green": 135, "blue": 35, "white": 10 },
+    // windows: rounding errors in 1) conversion to I420 (the capture),
+    // 2) downscaling, 3) conversion to RGB (for rendering).
+    "win": { "red": 5, "green": 5, "blue": 10, "white": 5 },
+    // linux: rounding errors in 1) conversion to I420 (the capture),
+    // 2) downscaling, 3) conversion to RGB (for rendering).
+    "linux": { "red": 5, "green": 5, "blue": 10, "white": 5 },
+    // android: we don't have a screen capture backend.
+    "android": { "red": 0, "green": 0, "blue": 0, "white": 0 },
+    // other: here just because it's supported by AppConstants.platform.
+    "other": { "red": 0, "green": 0, "blue": 0, "white": 0 },
+  };
+
+  const verifyScreenshare =
+      async (video, helper, upleft, upright, downleft, downright) => {
+    if (video.readyState < video.HAVE_CURRENT_DATA) {
+      info("Waiting for data");
+      await new Promise(r => video.onloadeddata = r);
+    }
+
+    // We assume video size will not change. Offsets help to account for a
+    // square fullscreen-canvas, while the screen is rectangular.
+    const offsetX = Math.max(0, video.videoWidth - video.videoHeight) / 2;
+    const offsetY = Math.max(0, video.videoHeight - video.videoWidth) / 2;
+
+    const verifyAround = async (internalX, internalY, color) => {
+      // Pick a couple of samples around a coordinate to check for a color.
+      // We check multiple rows and columns, to avoid most artifact issues.
+      let areaSamples = [
+        {dx: 0, dy: 0},
+        {dx: 1, dy: 3},
+        {dx: 8, dy: 5},
+      ];
+      const threshold = thresholds[AppConstants.platform][color.name];
+      for (let {dx, dy} of areaSamples) {
+        const x = offsetX + dx + internalX;
+        const y = offsetY + dy + internalY;
+        info(`Checking pixel (${[x,y]}) of total resolution `
+          + `${video.videoWidth}x${video.videoHeight} against ${color.name}.`);
+        let lastPixel = [-1, -1, -1, -1];
+        await helper.waitForPixel(video, px => {
+          lastPixel = Array.from(px);
+          return helper.isPixel(px, color, threshold);
+        }, {
+          offsetX: x,
+          offsetY: y,
+          cancel: wait(30000).then(_ =>
+            new Error(`Checking ${[x,y]} against ${color.name} timed out. ` +
+                      `Got [${lastPixel}]. Threshold ${threshold}.`)),
+        });
+        ok(true, `Pixel (${[x,y]}) passed. Got [${lastPixel}].`);
+      }
+    };
+
+    const screenSizeSq = Math.min(video.videoWidth, video.videoHeight);
+
+    info("Waiting for upper left quadrant to become " + upleft.name);
+    await verifyAround(screenSizeSq / 4, screenSizeSq / 4, upleft);
+
+    info("Waiting for upper right quadrant to become " + upright.name);
+    await verifyAround(screenSizeSq * 3 / 4, screenSizeSq / 4, upright);
+
+    info("Waiting for lower left quadrant to become " + downleft.name);
+    await verifyAround(screenSizeSq / 4, screenSizeSq * 3 / 4, downleft);
+
+    info("Waiting for lower right quadrant to become " + downright.name);
+    await verifyAround(screenSizeSq * 3 / 4, screenSizeSq * 3 / 4, downright);
+  };
+
+  /**
+   * Run a test to verify that we can complete a start and stop media playback
+   * cycle for a screenshare MediaStream on a video HTMLMediaElement.
+   */
+  runTest(async function () {
+    await pushPrefs(
+      ["full-screen-api.enabled", true],
+      ["full-screen-api.allow-trusted-requests-only", false],
+      ["full-screen-api.transition-duration.enter", "0 0"],
+      ["full-screen-api.transition-duration.leave", "0 0"],
+    );
+
+    // Improve real estate for screenshots
+    const test = document.getElementById("test");
+    test.setAttribute("style", "height:0;margin:0;");
+    const display = document.getElementById("display");
+    display.setAttribute("style", "margin:0;");
+    const testVideo = createMediaElement('video', 'testVideo');
+    testVideo.removeAttribute("width");
+    testVideo.removeAttribute("height");
+    testVideo.setAttribute("style", "max-height:240px;");
+
+    const canvas = document.createElement("canvas");
+    canvas.width = canvas.height = 20;
+    document.getElementById("content").appendChild(canvas);
+    const draw = ([upleft, upright, downleft, downright]) => {
+      helper.drawColor(canvas, helper[upleft], {offsetX: 0, offsetY: 0});
+      helper.drawColor(canvas, helper[upright], {offsetX: 10, offsetY: 0});
+      helper.drawColor(canvas, helper[downleft], {offsetX: 0, offsetY: 10});
+      helper.drawColor(canvas, helper[downright], {offsetX: 10, offsetY: 10});
+    };
+    const helper = new CaptureStreamTestHelper2D(1, 1);
+
+    const doVerify = async (stream, [upleft, upright, downleft, downright]) => {
+      // Reset from potential earlier verification runs.
+      testVideo.srcObject = null;
+      const playback = new MediaStreamPlayback(testVideo, stream);
+      playback.startMedia();
+      await playback.verifyPlaying();
+      const settings = stream.getTracks()[0].getSettings();
+      is(settings.width, testVideo.videoWidth,
+         "Width setting should match video width");
+      is(settings.height, testVideo.videoHeight,
+         "Height setting should match video height");
+      await SpecialPowers.wrap(canvas).requestFullscreen();
+      try {
+        await verifyScreenshare(testVideo, helper, helper[upleft], helper[upright],
+                                helper[downleft], helper[downright]);
+      } finally {
+        await playback.stopTracksForStreamInMediaPlayback();
+        await SpecialPowers.wrap(document).exitFullscreen();
+        // We wait a bit extra here to make sure we have completely left
+        // fullscreen when the --screenshot-on-fail screenshot is captured.
+        await wait(300);
+      }
+    };
+
+    info("Testing screenshare without constraints");
+    SpecialPowers.wrap(document).notifyUserGestureActivation();
+    let stream = await getUserMedia({video: {mediaSource: "screen"}});
+    let settings = stream.getTracks()[0].getSettings();
+    ok(settings.width <= 8192,
+       `Width setting ${settings.width} should be set after gUM (or 0 per bug 1453247)`);
+    ok(settings.height <= 8192,
+       `Height setting ${settings.height} should be set after gUM (or 0 per bug 1453247)`);
+    let colors = ["red", "blue", "green", "white"];
+    draw(colors);
+    await doVerify(stream, colors);
+    const screenWidth = testVideo.videoWidth;
+    const screenHeight = testVideo.videoHeight;
+
+    info("Testing screenshare with size and framerate constraints");
+    SpecialPowers.wrap(document).notifyUserGestureActivation();
+    for (const track of stream.getTracks()) {
+      track.stop();
+    }
+    stream = await getUserMedia({
+      video: {
+        mediaSource: 'screen',
+        width: {
+          min: '10',
+          max: '100'
+        },
+        height: {
+          min: '10',
+          max: '100'
+        },
+        frameRate: {
+          min: '10',
+          max: '15'
+        },
+      },
+    });
+    settings = stream.getTracks()[0].getSettings();
+    ok(settings.width == 0 || (settings.width >= 10 && settings.width <= 100),
+       `Width setting ${settings.width} should be correct after gUM (or 0 per bug 1453247)`);
+    ok(settings.height == 0 || (settings.height >= 10 && settings.height <= 100),
+       `Height setting ${settings.height} should be correct after gUM (or 0 per bug 1453247)`);
+    colors = ["green", "red", "white", "blue"];
+    draw(colors);
+    const streamClone = stream.clone();
+    await doVerify(streamClone, colors);
+    settings = stream.getTracks()[0].getSettings();
+    ok(settings.width >= 10 && settings.width <= 100,
+       `Width setting ${settings.width} should be within constraints`);
+    ok(settings.height >= 10 && settings.height <= 100,
+       `Height setting ${settings.height} should be within constraints`);
+    is(settings.width, testVideo.videoWidth,
+       "Width setting should match video width");
+    is(settings.height, testVideo.videoHeight,
+       "Height setting should match video height");
+    let expectedHeight = (screenHeight * settings.width) / screenWidth;
+    ok(Math.abs(expectedHeight - settings.height) <= 1,
+       "Aspect ratio after constrained gUM should be close enough");
+
+    info("Testing modifying screenshare with applyConstraints");
+    testVideo.srcObject = stream;
+    testVideo.play();
+    await new Promise(r => testVideo.onloadeddata = r);
+    const resize = haveEvent(
+      testVideo, "resize", wait(5000, new Error("Timeout waiting for resize")));
+    await stream.getVideoTracks()[0].applyConstraints({
+      mediaSource: 'screen',
+      width: 200,
+      height: 200,
+      frameRate: {
+        min: '5',
+        max: '10'
+      }
+    });
+    // getSettings() should report correct size as soon as applyConstraints()
+    // resolves - bug 1453259. Until fixed, check that we at least report
+    // something sane.
+    const newSettings = stream.getTracks()[0].getSettings();
+    ok(newSettings.width > settings.width && newSettings.width < screenWidth,
+       `Width setting ${newSettings.width} should have increased after applyConstraints`);
+    ok(newSettings.height > settings.height && newSettings.height < screenHeight,
+       `Height setting ${newSettings.height} should have increased after applyConstraints`);
+    await resize;
+    settings = stream.getTracks()[0].getSettings();
+    ok(settings.width > 100 && settings.width < screenWidth,
+       `Width setting ${settings.width} should have increased after first frame after applyConstraints`);
+    ok(settings.height > 100 && settings.height < screenHeight,
+       `Height setting ${settings.height} should have increased after first frame after applyConstraints`);
+    is(settings.width, testVideo.videoWidth,
+       "Width setting should match video width");
+    is(settings.height, testVideo.videoHeight,
+       "Height setting should match video height");
+    expectedHeight = (screenHeight * settings.width) / screenWidth;
+    ok(Math.abs(expectedHeight - settings.height) <= 1,
+       "Aspect ratio after applying constraints should be close enough");
+    colors = ["white", "green", "blue", "red"];
+    draw(colors);
+    await doVerify(stream, colors);
+    for (const track of [...stream.getTracks(), ...streamClone.getTracks()]) {
+      track.stop();
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicTabshare.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicTabshare.html
new file mode 100644
index 0000000000..635cf387d4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicTabshare.html
@@ -0,0 +1,67 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia Basic Tabshare Test",
+    bug: "1193075"
+  });
+  /**
+   * Run a test to verify that we can complete a start and stop media playback
+   * cycle for a tabshare MediaStream on a video HTMLMediaElement.
+   *
+   * Additionally, exercise applyConstraints code for tabshare viewport offset.
+   */
+  runTest(function () {
+    var testVideo = createMediaElement('video', 'testVideo');
+
+    return Promise.resolve()
+      .then(() => pushPrefs(["media.getusermedia.browser.enabled", true]))
+      .then(() => {
+        SpecialPowers.wrap(document).notifyUserGestureActivation();
+        return getUserMedia({
+          video: { mediaSource: "browser",
+                   scrollWithPage: true },
+        });
+      })
+      .then(stream => {
+        var playback = new MediaStreamPlayback(testVideo, stream);
+        return playback.playMedia(false);
+      })
+      .then(() => getUserMedia({
+        video: {
+          mediaSource: "browser",
+          viewportOffsetX: 0,
+          viewportOffsetY: 0,
+          viewportWidth: 100,
+          viewportHeight: 100
+        },
+      }))
+      .then(stream => {
+        var playback = new MediaStreamPlayback(testVideo, stream);
+        playback.startMedia(false);
+        return playback.verifyPlaying()
+          .then(() => Promise.all([
+            () => testVideo.srcObject.getVideoTracks()[0].applyConstraints({
+              mediaSource: "browser",
+              viewportOffsetX: 10,
+              viewportOffsetY: 50,
+              viewportWidth: 90,
+              viewportHeight: 50
+            }),
+            () => listenUntil(testVideo, "resize", () => true)
+          ]))
+          .then(() => playback.verifyPlaying()) // still playing
+          .then(() => playback.stopTracksForStreamInMediaPlayback())
+          .then(() => playback.detachFromMediaElement());
+      });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideo.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideo.html
new file mode 100644
index 0000000000..786d9f2e4b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideo.html
@@ -0,0 +1,30 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia Basic Video Test",
+    bug: "781534"
+  });
+  /**
+   * Run a test to verify that we can complete a start and stop media playback
+   * cycle for an video MediaStream on a video HTMLMediaElement.
+   */
+  runTest(function () {
+    var testVideo = createMediaElement('video', 'testVideo');
+    var constraints = {video: true};
+
+    return getUserMedia(constraints).then(stream => {
+      var playback = new MediaStreamPlayback(testVideo, stream);
+      return playback.playMedia(false);
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideoAudio.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideoAudio.html
new file mode 100644
index 0000000000..5218bf7301
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideoAudio.html
@@ -0,0 +1,30 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia Basic Video & Audio Test",
+    bug: "781534"
+  });
+  /**
+   * Run a test to verify that we can complete a start and stop media playback
+   * cycle for a video and audio MediaStream on a video HTMLMediaElement.
+   */
+  runTest(function () {
+    var testVideoAudio = createMediaElement('video', 'testVideoAudio');
+    var constraints = {video: true, audio: true};
+
+    return getUserMedia(constraints).then(stream => {
+      var playback = new MediaStreamPlayback(testVideoAudio, stream);
+      return playback.playMedia(false);
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideo_playAfterLoadedmetadata.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideo_playAfterLoadedmetadata.html
new file mode 100644
index 0000000000..fbab1b4357
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideo_playAfterLoadedmetadata.html
@@ -0,0 +1,42 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia Basic Video shall receive 'loadedmetadata' without play()ing",
+    bug: "1149494"
+  });
+  /**
+   * Run a test to verify that we will always get 'loadedmetadata' from a video
+   * HTMLMediaElement playing a gUM MediaStream.
+   */
+  runTest(() => {
+    var testVideo = createMediaElement('video', 'testVideo');
+    var constraints = {video: true};
+
+    return getUserMedia(constraints).then(stream => {
+      var playback = new MediaStreamPlayback(testVideo, stream);
+      var video = playback.mediaElement;
+
+      video.srcObject = stream;
+      return new Promise(resolve => {
+        ok(playback.mediaElement.paused,
+           "Media element should be paused before play()ing");
+        video.addEventListener('loadedmetadata', function() {
+          ok(video.videoWidth > 0, "Expected nonzero video width");
+          ok(video.videoHeight > 0, "Expected nonzero video width");
+          resolve();
+        });
+      })
+      .then(() => stream.getTracks().forEach(t => t.stop()));
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicWindowshare.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicWindowshare.html
new file mode 100644
index 0000000000..7b27944bdc
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicWindowshare.html
@@ -0,0 +1,39 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia Basic Windowshare Test",
+    bug: "1038926"
+  });
+  /**
+   * Run a test to verify that we can complete a start and stop media playback
+   * cycle for an screenshare MediaStream on a video HTMLMediaElement.
+   */
+  runTest(async function () {
+    const testVideo = createMediaElement('video', 'testVideo');
+    const constraints = {
+      video: { mediaSource: "window" },
+    };
+
+    try {
+      await getUserMedia(constraints);
+      ok(false, "Should require user gesture");
+    } catch (e) {
+      is(e.name, "InvalidStateError");
+    }
+
+    SpecialPowers.wrap(document).notifyUserGestureActivation();
+    const stream = await getUserMedia(constraints);
+    const playback = new MediaStreamPlayback(testVideo, stream);
+    return playback.playMedia(false);
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_bug1223696.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_bug1223696.html
new file mode 100644
index 0000000000..6af7b69d70
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_bug1223696.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  "use strict";
+
+  createHTML({
+    title: "Testing that removeTrack+addTrack of video tracks still render the correct track in a media element",
+    bug: "1223696",
+    visible: true
+  });
+
+  runTest(async function() {
+    const stream = await getUserMedia({audio:true, video: true});
+    info("Test addTrack()ing a video track to an audio-only gUM stream");
+
+    const video = createMediaElement("video", "test_video_track");
+    video.srcObject = stream;
+    video.play();
+
+    await haveEvent(video, "loadeddata", wait(5000, new Error("Timeout")));
+    info("loadeddata");
+
+    const removedTrack = stream.getVideoTracks()[0];
+    stream.removeTrack(removedTrack);
+
+    const h = new CaptureStreamTestHelper2D();
+    const emitter = new VideoFrameEmitter(h.grey, h.grey);
+    emitter.start();
+
+    stream.addTrack(emitter.stream().getVideoTracks()[0]);
+
+    checkMediaStreamContains(stream, [stream.getAudioTracks()[0],
+                                      emitter.stream().getVideoTracks()[0]]);
+
+    await h.pixelMustBecome(video, h.grey, {
+      threshold: 5,
+      infoString: "The canvas track should be rendered by the media element",
+    });
+
+    emitter.stop();
+    for (const t of [removedTrack, ...stream.getAudioTracks()]) {
+      t.stop();
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_callbacks.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_callbacks.html
new file mode 100644
index 0000000000..14c6cc7e7f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_callbacks.html
@@ -0,0 +1,35 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "navigator.mozGetUserMedia Callback Test",
+    bug: "1119593"
+  });
+  /**
+   * Check that the old fashioned callback-based function works.
+   */
+  runTest(function () {
+    var testAudio = createMediaElement('audio', 'testAudio');
+    var constraints = {audio: true};
+
+    SimpleTest.waitForExplicitFinish();
+    return new Promise(resolve =>
+      navigator.mozGetUserMedia(constraints, stream => {
+        checkMediaStreamTracks(constraints, stream);
+
+        var playback = new MediaStreamPlayback(testAudio, stream);
+        return playback.playMedia(false)
+          .then(() => resolve(), generateErrorCallback());
+      }, generateErrorCallback())
+    );
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_constraints.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_constraints.html
new file mode 100644
index 0000000000..d6439ce9d6
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_constraints.html
@@ -0,0 +1,166 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+  <script src="constraints.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({ title: "Test getUserMedia constraints", bug: "882145" });
+/**
+  Tests covering gUM constraints API for audio, video and fake video. Exercise
+  successful parsing code and ensure that unknown required constraints and
+  overconstraining cases produce appropriate errors.
+*/
+var tests = [
+  // Each test here tests a different constraint or codepath.
+  { message: "unknown required constraint on video ignored",
+    constraints: { video: { somethingUnknown: { exact: 0 } } },
+    error: null },
+  { message: "unknown required constraint on audio ignored",
+    constraints: { audio: { somethingUnknown: { exact: 0 } } },
+    error: null },
+  { message: "audio overconstrained by facingMode ignored",
+    constraints: { audio: { facingMode: { exact: 'left' } } },
+    error: null },
+  { message: "full screensharing requires permission",
+    constraints: { video: { mediaSource: 'screen' } },
+    error: "NotAllowedError" },
+  { message: "application screensharing no longer exists",
+    constraints: { video: { mediaSource: 'application' } },
+    error: "OverconstrainedError" },
+  { message: "window screensharing requires permission",
+    constraints: { video: { mediaSource: 'window' } },
+    error: "NotAllowedError" },
+  { message: "browser screensharing requires permission",
+    constraints: { video: { mediaSource: 'browser' } },
+    error: "NotAllowedError" },
+  { message: "unknown mediaSource in video fails",
+    constraints: { video: { mediaSource: 'uncle' } },
+    error: "OverconstrainedError",
+    constraint: "mediaSource" },
+  { message: "unknown mediaSource in audio fails",
+    constraints: { audio: { mediaSource: 'uncle' } },
+    error: "OverconstrainedError",
+    constraint: "mediaSource" },
+  { message: "emtpy constraint fails",
+    constraints: { },
+    error: "TypeError" },
+  { message: "Triggering mock failure in default video device fails",
+    constraints: { video: { deviceId: 'bad device' } },
+    error: "NotReadableError" },
+  { message: "Triggering mock failure in default audio device fails",
+    constraints: { audio: { deviceId: 'bad device' } },
+    error: "NotReadableError" },
+  { message: "Success-path: optional video facingMode + audio ignoring facingMode",
+    constraints: { audio: { mediaSource: 'microphone',
+                            facingMode: 'left',
+                            foo: 0,
+                            advanced: [{ facingMode: 'environment' },
+                                       { facingMode: 'user' },
+                                       { bar: 0 }] },
+                   video: { mediaSource: 'camera',
+                            foo: 0,
+                            advanced: [{ facingMode: 'environment' },
+                                       { facingMode: ['user'] },
+                                       { facingMode: ['left', 'right', 'user'] },
+                                       { bar: 0 }] } },
+    error: null },
+  { message: "legacy facingMode ignored",
+    constraints: { video: { mandatory: { facingMode: 'left' } } },
+    error: null },
+];
+
+var mustSupport = [
+  'width', 'height', 'frameRate', 'facingMode', 'deviceId', 'groupId',
+  'echoCancellation', 'noiseSuppression', 'autoGainControl', 'channelCount',
+
+  // Yet to add:
+  //  'aspectRatio', 'volume', 'sampleRate', 'sampleSize', 'latency'
+
+  // http://fluffy.github.io/w3c-screen-share/#screen-based-video-constraints
+  // OBE by http://w3c.github.io/mediacapture-screen-share
+  'mediaSource',
+
+  // Experimental https://bugzilla.mozilla.org/show_bug.cgi?id=1131568#c3
+  'browserWindow', 'scrollWithPage',
+  'viewportOffsetX', 'viewportOffsetY', 'viewportWidth', 'viewportHeight',
+];
+
+var mustFailWith = (msg, reason, constraint, f) =>
+  f().then(() => ok(false, msg + " must fail"), e => {
+    is(e.name, reason, msg + " must fail: " + e.message);
+    if (constraint !== undefined) {
+      is(e.constraint, constraint, msg + " must fail w/correct constraint.");
+    }
+  });
+
+/**
+ * Starts the test run by running through each constraint
+ * test by verifying that the right resolution and rejection is fired.
+ */
+
+runTest(() => pushPrefs(
+    // This test expects fake devices, particularly for the 'triggering mock
+    // failure *' steps. So explicitly disable loopback and setup fakes
+    ['media.audio_loopback_dev', ''],
+    ['media.video_loopback_dev', ''],
+    ['media.navigator.streams.fake', true]
+  )
+  .then(() => {
+    // Check supported constraints first.
+    var dict = navigator.mediaDevices.getSupportedConstraints();
+    var supported = Object.keys(dict);
+
+    mustSupport.forEach(key => ok(supported.includes(key) && dict[key],
+                                  "Supports " + key));
+
+    var unexpected = supported.filter(key => !mustSupport.includes(key));
+    is(unexpected.length, 0,
+       "Unanticipated support (please update test): " + unexpected);
+  })
+  .then(() => pushPrefs(["media.getusermedia.browser.enabled", false],
+                        ["media.getusermedia.screensharing.enabled", false]))
+  .then(() => tests.reduce((p, test) => p.then(
+    () => {
+      SpecialPowers.wrap(document).notifyUserGestureActivation();
+      return getUserMedia(test.constraints);
+    })
+    .then(stream => {
+      is(null, test.error, test.message);
+      stream.getTracks().forEach(t => t.stop());
+    }, e => {
+      is(e.name, test.error, test.message + ": " + e.message);
+      if (test.constraint) {
+        is(e.constraint, test.constraint,
+           test.message + " w/correct constraint.");
+      }
+    }), Promise.resolve()))
+  .then(() => getUserMedia({video: true, audio: true}))
+  .then(stream => stream.getVideoTracks()[0].applyConstraints({ width: 320 })
+    .then(() => stream.getAudioTracks()[0].applyConstraints({ }))
+    .then(() => {
+      stream.getTracks().forEach(track => track.stop());
+      ok(true, "applyConstraints code exercised");
+    }))
+  // TODO: Test outcome once fake devices support constraints (Bug 1088621)
+  .then(() => mustFailWith("applyConstraints fails on non-Gum tracks",
+                           "OverconstrainedError", "",
+                           () => (new AudioContext())
+                               .createMediaStreamDestination().stream
+                               .getAudioTracks()[0].applyConstraints()))
+  .then(() => mustFailWith(
+      "getUserMedia with unsatisfied required constraint",
+      "OverconstrainedError", "deviceId",
+      () => getUserMedia({ audio: true,
+                           video: { deviceId: { exact: "unheardof" } } })))
+  .then(() => mustFailWith(
+      "getUserMedia with unsatisfied required constraint array",
+      "OverconstrainedError", "deviceId",
+      () => getUserMedia({ audio: true,
+                           video: { deviceId: { exact: ["a", "b"] } } }))));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_cubebDisabled.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_cubebDisabled.html
new file mode 100644
index 0000000000..54142aeb77
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_cubebDisabled.html
@@ -0,0 +1,42 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia with Cubeb Disabled Test",
+    bug: "1443525"
+  });
+  /**
+   * Run a test to verify we fail gracefully if we cannot fetch a cubeb context
+   * during a gUM call.
+   */
+  runTest(async function () {
+    info("Get user media with cubeb disabled starting");
+    // Push prefs to ensure no cubeb context and no fake streams.
+    await pushPrefs(["media.cubeb.force_null_context", true],
+                    ["media.navigator.permission.device", false],
+                    ["media.navigator.streams.fake", false]);
+
+    // Request audio only, to avoid cams
+    let constraints = {audio: true, video: false};
+    let stream;
+    try {
+      stream = await getUserMedia(constraints);
+    } catch (e) {
+      // We've got no audio backend, so we expect gUM to fail.
+      ok(e.name == "NotFoundError", "Expected NotFoundError due to no audio tracks!");
+      return;
+    }
+    // If we're not on android we should not have gotten a stream without a cubeb context!
+    ok(false, "getUserMedia not expected to succeed when cubeb is disabled, but it did!");
+  });
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_cubebDisabledFakeStreams.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_cubebDisabledFakeStreams.html
new file mode 100644
index 0000000000..f8150cc4c1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_cubebDisabledFakeStreams.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia fake stream with Cubeb Disabled Test",
+    bug: "1443525"
+  });
+  /**
+   * Run a test to verify we can still return a fake stream even if we cannot
+   * get a cubeb context. See also Bug 1434477
+   */
+  runTest(async function () {
+    info("Get user media with cubeb disabled and fake tracks starting");
+    // Push prefs to ensure no cubeb context and fake streams
+    await pushPrefs(["media.cubeb.force_null_context", true],
+                    ["media.navigator.streams.fake", true],
+                    ['media.audio_loopback_dev', '']);
+    let testAudio = createMediaElement('audio', 'testAudio');
+    // Request audio only, to avoid cams
+    let constraints = {audio: true, video: false};
+    let stream;
+    try {
+      stream = await getUserMedia(constraints);
+    } catch (e) {
+      // We've got no audio backend, so we expect gUM to fail
+      ok(false, `Did not expect to fail, but got ${e}`);
+      return;
+    }
+    ok(stream, "getUserMedia should get a stream!");
+    let playback = new MediaStreamPlayback(testAudio, stream);
+    return playback.playMedia(false);
+  });
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_getTrackById.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_getTrackById.html
new file mode 100644
index 0000000000..161bf631e3
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_getTrackById.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "Basic getTrackById test of gUM stream",
+    bug: "1208390",
+  });
+
+  runTest(() => {
+    var constraints = {audio: true, video: true};
+    return getUserMedia(constraints).then(stream => {
+      is(stream.getTrackById(""), null,
+         "getTrackById of non-matching string should return null");
+
+      let audioTrack = stream.getAudioTracks()[0];
+      is(stream.getTrackById(audioTrack.id), audioTrack,
+         "getTrackById with matching id should return the track");
+
+      let videoTrack = stream.getVideoTracks()[0];
+      is(stream.getTrackById(videoTrack.id), videoTrack,
+         "getTrackById with matching id should return the track");
+
+      stream.removeTrack(audioTrack);
+      is(stream.getTrackById(audioTrack.id), null,
+         "getTrackById with id of removed track should return null");
+
+      let newStream = new MediaStream();
+      is(newStream.getTrackById(videoTrack.id), null,
+         "getTrackById with id of track in other stream should return null");
+
+      newStream.addTrack(audioTrack);
+      is(newStream.getTrackById(audioTrack.id), audioTrack,
+         "getTrackByid with matching id should return the track");
+
+      newStream.addTrack(videoTrack);
+      is(newStream.getTrackById(videoTrack.id), videoTrack,
+         "getTrackByid with matching id should return the track");
+      [audioTrack, videoTrack].forEach(t => t.stop());
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_gumWithinGum.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_gumWithinGum.html
new file mode 100644
index 0000000000..86a7aa5606
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_gumWithinGum.html
@@ -0,0 +1,38 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({title: "getUserMedia within getUserMedia", bug: "822109" });
+  /**
+   * Run a test that we can complete a playback cycle for a video,
+   * then upon completion, do a playback cycle with audio, such that
+   * the audio gum call happens within the video gum call.
+   */
+  runTest(function () {
+    return getUserMedia({video: true})
+      .then(videoStream => {
+        var testVideo = createMediaElement('video', 'testVideo');
+        var videoPlayback = new MediaStreamPlayback(testVideo,
+                                                    videoStream);
+
+        return videoPlayback.playMediaWithoutStoppingTracks(false)
+          .then(() => getUserMedia({audio: true}))
+          .then(audioStream => {
+            var testAudio = createMediaElement('audio', 'testAudio');
+            var audioPlayback = new MediaStreamPlayback(testAudio,
+                                                        audioStream);
+
+            return audioPlayback.playMedia(false);
+          })
+          .then(() => videoStream.getTracks().forEach(t => t.stop()));
+      });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_loadedmetadata.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_loadedmetadata.html
new file mode 100644
index 0000000000..d6efac4650
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_loadedmetadata.html
@@ -0,0 +1,39 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia in media element should have video dimensions on loadedmetadata",
+    bug: "1240478"
+  });
+  /**
+   * Tests that assigning a stream to a media element results in the
+   * "loadedmetadata" event without having to play() the media element.
+   *
+   * Also makes sure that the video size has been set on "loadedmetadata".
+   */
+  runTest(function () {
+    var v = document.createElement("video");
+    document.body.appendChild(v);
+    v.preload = "metadata";
+
+    var constraints = {video: true, audio: true};
+    return getUserMedia(constraints).then(stream => new Promise(resolve => {
+      v.srcObject = stream;
+      v.onloadedmetadata = () => {
+        isnot(v.videoWidth, 0, "videoWidth shall be set on 'loadedmetadata'");
+        isnot(v.videoHeight, 0, "videoHeight shall be set on 'loadedmetadata'");
+        resolve();
+      };
+    })
+    .then(() => stream.getTracks().forEach(t => t.stop())));
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_audio.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_audio.html
new file mode 100644
index 0000000000..3b9e00896c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_audio.html
@@ -0,0 +1,116 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script>
+
+createHTML({
+  bug: "1259788",
+  title: "Test CaptureStream audio content on HTMLMediaElement playing a gUM MediaStream",
+  visible: true
+});
+
+var audioContext;
+var gUMAudioElement;
+var analyser;
+runTest(() => getUserMedia({audio: { echoCancellation: false }})
+  .then(stream => {
+    gUMAudioElement = createMediaElement("audio", "gUMAudio");
+    gUMAudioElement.srcObject = stream;
+
+    audioContext = new AudioContext();
+    info("Capturing");
+
+    analyser = new AudioStreamAnalyser(audioContext,
+                                       gUMAudioElement.mozCaptureStream());
+    analyser.enableDebugCanvas();
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] > 200 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50);
+  })
+  .then(() => {
+    info("Audio flowing. Pausing.");
+    gUMAudioElement.pause();
+
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] < 50 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50);
+  })
+  .then(() => {
+    info("Audio stopped flowing. Playing.");
+    gUMAudioElement.play();
+
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] > 200 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50);
+  })
+  .then(() => {
+    info("Audio flowing. Removing source.");
+    var stream = gUMAudioElement.srcObject;
+    gUMAudioElement.srcObject = null;
+
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] < 50 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50)
+        .then(() => stream);
+  })
+  .then(stream => {
+    info("Audio stopped flowing. Setting source.");
+    gUMAudioElement.srcObject = stream;
+
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] > 200 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50);
+  })
+  .then(() => {
+    info("Audio flowing from new source. Adding a track.");
+    let oscillator = audioContext.createOscillator();
+    oscillator.type = 'sine';
+    oscillator.frequency.value = 2000;
+    oscillator.start();
+
+    let oscOut = audioContext.createMediaStreamDestination();
+    oscillator.connect(oscOut);
+
+    gUMAudioElement.srcObject.addTrack(oscOut.stream.getTracks()[0]);
+
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] > 200 &&
+      array[analyser.binIndexForFrequency(1500)]            < 50 &&
+      array[analyser.binIndexForFrequency(2000)]            > 200 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50);
+  })
+  .then(() => {
+    info("Audio flowing from new track. Removing a track.");
+
+    const gUMTrack = gUMAudioElement.srcObject.getTracks()[0];
+    gUMAudioElement.srcObject.removeTrack(gUMTrack);
+
+    is(gUMAudioElement.srcObject.getTracks().length, 1,
+       "A track should have been removed");
+
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] < 50 &&
+      array[analyser.binIndexForFrequency(1500)]            < 50 &&
+      array[analyser.binIndexForFrequency(2000)]            > 200 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50)
+        .then(() => [gUMTrack, ...gUMAudioElement.srcObject.getTracks()]
+            .forEach(t => t.stop()));
+  })
+  .then(() => ok(true, "Test passed."))
+  .catch(e => ok(false, "Test failed: " + e + (e.stack ? "\n" + e.stack : ""))));
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_tracks.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_tracks.html
new file mode 100644
index 0000000000..a747e75de9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_tracks.html
@@ -0,0 +1,179 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script>
+
+createHTML({
+  bug: "1259788",
+  title: "Test CaptureStream track output on HTMLMediaElement playing a gUM MediaStream",
+  visible: true
+});
+
+let audioElement;
+let audioCaptureStream;
+let videoElement;
+let videoCaptureStream;
+let untilEndedElement;
+let streamUntilEnded;
+const tracks = [];
+runTest(async () => {
+  try {
+    let stream = await getUserMedia({audio: true, video: true});
+    // We need to test with multiple tracks. We add an extra of each kind.
+    for (const track of stream.getTracks()) {
+      stream.addTrack(track.clone());
+    }
+
+    audioElement = createMediaElement("audio", "gUMAudio");
+    audioElement.srcObject = stream;
+
+    await haveEvent(audioElement, "loadedmetadata", wait(50000, new Error("Timeout")));
+
+    info("Capturing audio element (loadedmetadata -> captureStream)");
+    audioCaptureStream = audioElement.mozCaptureStream();
+
+    is(audioCaptureStream.getAudioTracks().length, 2,
+      "audio element should capture two audio tracks");
+    is(audioCaptureStream.getVideoTracks().length, 0,
+      "audio element should not capture any video tracks");
+
+    await haveNoEvent(audioCaptureStream, "addtrack");
+
+    videoElement = createMediaElement("video", "gUMVideo");
+
+    info("Capturing video element (captureStream -> loadedmetadata)");
+    videoCaptureStream = videoElement.mozCaptureStream();
+    videoElement.srcObject = audioElement.srcObject.clone();
+
+    is(videoCaptureStream.getTracks().length, 0,
+      "video element should have no tracks before metadata known");
+
+    await haveEventsButNoMore(
+      videoCaptureStream, "addtrack", 3, wait(50000, new Error("No event")));
+
+    is(videoCaptureStream.getAudioTracks().length, 2,
+      "video element should capture two audio tracks");
+    is(videoCaptureStream.getVideoTracks().length, 1,
+      "video element should capture one video track at most");
+
+    info("Testing dynamically adding audio track to audio element");
+    audioElement.srcObject.addTrack(
+      audioElement.srcObject.getAudioTracks()[0].clone());
+    await haveEventsButNoMore(
+      audioCaptureStream, "addtrack", 1, wait(50000, new Error("No event")));
+
+    is(audioCaptureStream.getAudioTracks().length, 3,
+      "Audio element should have three audio tracks captured.");
+
+    info("Testing dynamically adding video track to audio element");
+    audioElement.srcObject.addTrack(
+      audioElement.srcObject.getVideoTracks()[0].clone());
+    await haveNoEvent(audioCaptureStream, "addtrack");
+
+    is(audioCaptureStream.getVideoTracks().length, 0,
+      "Audio element should have no video tracks captured.");
+
+    info("Testing dynamically adding audio track to video element");
+    videoElement.srcObject.addTrack(
+      videoElement.srcObject.getAudioTracks()[0].clone());
+    await haveEventsButNoMore(
+      videoCaptureStream, "addtrack", 1, wait(50000, new Error("Timeout")));
+
+    is(videoCaptureStream.getAudioTracks().length, 3,
+      "Captured video stream should have three audio tracks captured.");
+
+    info("Testing dynamically adding video track to video element");
+    videoElement.srcObject.addTrack(
+      videoElement.srcObject.getVideoTracks()[0].clone());
+    await haveNoEvent(videoCaptureStream, "addtrack");
+
+    is(videoCaptureStream.getVideoTracks().length, 1,
+      "Captured video stream should have at most one video tracks captured.");
+
+    info("Testing track removal.");
+    tracks.push(...videoElement.srcObject.getTracks());
+    for (const track of videoElement.srcObject.getVideoTracks().reverse()) {
+      videoElement.srcObject.removeTrack(track);
+    }
+    is(videoCaptureStream.getVideoTracks().length, 1,
+      "Captured video should have still have one video track.");
+
+    await haveEvent(videoCaptureStream.getVideoTracks()[0], "ended",
+      wait(50000, new Error("Timeout")));
+    await haveEvent(videoCaptureStream, "removetrack",
+      wait(50000, new Error("Timeout")));
+
+    is(videoCaptureStream.getVideoTracks().length, 0,
+      "Captured video stream should have no video tracks after removal.");
+
+
+    info("Testing source reset.");
+    stream = await getUserMedia({audio: true, video: true});
+    videoElement.srcObject = stream;
+    for (const track of videoCaptureStream.getTracks()) {
+      await Promise.race(videoCaptureStream.getTracks().map(
+        t => haveEvent(t, "ended", wait(50000, new Error("Timeout"))))
+      );
+      await haveEvent(videoCaptureStream, "removetrack", wait(50000, new Error("Timeout")));
+    }
+    await haveEventsButNoMore(
+      videoCaptureStream, "addtrack", 2, wait(50000, new Error("Timeout")));
+    is(videoCaptureStream.getAudioTracks().length, 1,
+      "Captured video stream should have one audio track");
+
+    is(videoCaptureStream.getVideoTracks().length, 1,
+      "Captured video stream should have one video track");
+
+    info("Testing CaptureStreamUntilEnded");
+    untilEndedElement =
+      createMediaElement("video", "gUMVideoUntilEnded");
+    untilEndedElement.srcObject = audioElement.srcObject;
+
+    await haveEvent(untilEndedElement, "loadedmetadata",
+      wait(50000, new Error("Timeout")));
+
+    streamUntilEnded = untilEndedElement.mozCaptureStreamUntilEnded();
+
+    is(streamUntilEnded.getAudioTracks().length, 3,
+      "video element should capture all 3 audio tracks until ended");
+    is(streamUntilEnded.getVideoTracks().length, 1,
+      "video element should capture only 1 video track until ended");
+
+    for (const track of untilEndedElement.srcObject.getTracks()) {
+      track.stop();
+    }
+
+    await haveEvent(untilEndedElement, "ended", wait(50000, new Error("Timeout")));
+    for (const track of streamUntilEnded.getTracks()) {
+      await Promise.race(streamUntilEnded.getTracks().map(
+        t => haveEvent(t, "ended", wait(50000, new Error("Timeout"))))
+      );
+      await haveEvent(streamUntilEnded, "removetrack", wait(50000, new Error("Timeout")));
+    }
+
+    info("Element and tracks ended. Ensuring that new tracks aren't created.");
+    untilEndedElement.srcObject = videoElement.srcObject;
+    await haveEventsButNoMore(
+      untilEndedElement, "loadedmetadata", 1, wait(50000, new Error("Timeout")));
+
+    is(streamUntilEnded.getTracks().length, 0, "Should have no tracks");
+  } catch(e) {
+    ok(false, "Test failed: " + e + (e && e.stack ? "\n" + e.stack : ""));
+  } finally {
+    if (videoElement) {
+      tracks.push(...videoElement.srcObject.getTracks());
+    }
+    for(const track of tracks) {
+      track.stop();
+    }
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_video.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_video.html
new file mode 100644
index 0000000000..d177e93bfb
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_video.html
@@ -0,0 +1,91 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script>
+
+createHTML({
+  bug: "1259788",
+  title: "Test CaptureStream video content on HTMLMediaElement playing a gUM MediaStream",
+  visible: true
+});
+
+var gUMVideoElement;
+var captureStreamElement;
+
+const pausedTimeout = 1000;
+let h;
+
+runTest(async () => {
+  try {
+    await pushPrefs(
+      // This test expects fake video devices, as it expects captured frames to
+      // shift over time, which is not currently provided by loopback devices
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+
+    let stream = await getUserMedia({video: true});
+    h = new VideoStreamHelper();
+    gUMVideoElement =
+      createMediaElement("video", "gUMVideo");
+    gUMVideoElement.srcObject = stream;
+    gUMVideoElement.play();
+
+    info("Capturing");
+    captureStreamElement =
+      createMediaElement("video", "captureStream");
+    captureStreamElement.srcObject = gUMVideoElement.mozCaptureStream();
+    captureStreamElement.play();
+
+    await h.checkVideoPlaying(captureStreamElement);
+
+    // Adding a dummy audio track to the stream will keep a consuming media
+    // element from ending.
+    // We could also solve it by repeatedly play()ing or autoplay, but then we
+    // wouldn't be sure the media element stopped rendering video because it
+    // went to the ended state or because there were no frames for the track.
+    let osc = createOscillatorStream(new AudioContext(), 1000);
+    captureStreamElement.srcObject.addTrack(osc.getTracks()[0]);
+
+    info("Video flowing. Pausing.");
+    gUMVideoElement.pause();
+    await h.checkVideoPaused(captureStreamElement, { time: pausedTimeout });
+
+    info("Video stopped flowing. Playing.");
+    gUMVideoElement.play();
+    await h.checkVideoPlaying(captureStreamElement);
+
+    info("Video flowing. Removing source.");
+    stream = gUMVideoElement.srcObject;
+    gUMVideoElement.srcObject = null;
+    await h.checkVideoPaused(captureStreamElement, { time: pausedTimeout });
+
+    info("Video stopped flowing. Setting source.");
+    gUMVideoElement.srcObject = stream;
+    await h.checkVideoPlaying(captureStreamElement);
+
+    info("Video flowing. Changing source by track manipulation. Remove first.");
+    let track = gUMVideoElement.srcObject.getTracks()[0];
+    gUMVideoElement.srcObject.removeTrack(track);
+    await h.checkVideoPaused(captureStreamElement, { time: pausedTimeout });
+
+    info("Video paused. Changing source by track manipulation. Add first.");
+    gUMVideoElement.srcObject.addTrack(track);
+    gUMVideoElement.play();
+    await h.checkVideoPlaying(captureStreamElement);
+
+    gUMVideoElement.srcObject.getTracks().forEach(t => t.stop());
+    ok(true, "Test passed.");
+  } catch (e) {
+    ok(false, "Test failed: " + e + (e.stack ? "\n" + e.stack : ""));
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamClone.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamClone.html
new file mode 100644
index 0000000000..029ce77dd0
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamClone.html
@@ -0,0 +1,258 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "MediaStream.clone()",
+  bug: "1208371"
+});
+
+runTest(async () => {
+  await pushPrefs(
+    ["media.getusermedia.camera.stop_on_disable.enabled", true],
+    ["media.getusermedia.camera.stop_on_disable.delay_ms", 0],
+    ["media.getusermedia.microphone.stop_on_disable.enabled", true],
+    ["media.getusermedia.microphone.stop_on_disable.delay_ms", 0]);
+
+  let gUMStream = await getUserMedia({audio: true, video: true});
+  {
+    info("Test clone()ing an audio/video gUM stream");
+    let clone = gUMStream.clone();
+
+    checkMediaStreamCloneAgainstOriginal(clone, gUMStream);
+    checkMediaStreamTrackCloneAgainstOriginal(clone.getAudioTracks()[0],
+                                              gUMStream.getAudioTracks()[0]);
+    checkMediaStreamTrackCloneAgainstOriginal(clone.getVideoTracks()[0],
+                                              gUMStream.getVideoTracks()[0]);
+
+    isnot(clone.id.length, 0, "Stream clone should have an id string");
+    isnot(clone.getAudioTracks()[0].id.length, 0,
+          "Audio track clone should have an id string");
+    isnot(clone.getVideoTracks()[0].id.length, 0,
+          "Audio track clone should have an id string");
+
+    info("Playing from track clones");
+    let test = createMediaElement('video', 'testClonePlayback');
+    let playback = new MediaStreamPlayback(test, clone);
+    await playback.playMedia(false);
+  }
+
+  {
+    info("Test addTrack()ing a video track to a stream without affecting its clone");
+    let stream = new MediaStream(gUMStream.getVideoTracks());
+    let otherStream = await getUserMedia({video: true});
+    let track = stream.getTracks()[0];
+    let otherTrack = otherStream.getTracks()[0];
+
+    let streamClone = stream.clone();
+    let trackClone = streamClone.getTracks()[0];
+    checkMediaStreamContains(streamClone, [trackClone], "Initial clone");
+
+    stream.addTrack(otherTrack);
+    checkMediaStreamContains(stream, [track, otherTrack],
+                             "Added video to original");
+    checkMediaStreamContains(streamClone, [trackClone],
+                             "Clone not affected");
+
+    stream.removeTrack(track);
+    streamClone.addTrack(track);
+    checkMediaStreamContains(streamClone, [trackClone, track],
+                             "Added video to clone");
+    checkMediaStreamContains(stream, [otherTrack],
+                             "Original not affected");
+
+    // Not part of streamClone. Does not get stopped by the playback test.
+    otherTrack.stop();
+
+    let test = createMediaElement('video', 'testClonePlayback');
+    let playback = new MediaStreamPlayback(test, streamClone);
+    await playback.playMedia(false);
+  }
+
+  {
+    info("Test cloning a stream into inception");
+    let stream = gUMStream.clone()
+    let clone = stream;
+    let clones = Array(10).fill().map(() => clone = clone.clone());
+    let inceptionClone = clones.pop();
+    checkMediaStreamCloneAgainstOriginal(inceptionClone, stream);
+    stream.getTracks().forEach(t => (stream.removeTrack(t),
+                                     inceptionClone.addTrack(t)));
+    is(inceptionClone.getAudioTracks().length, 2,
+       "The inception clone should contain the original audio track and a track clone");
+    is(inceptionClone.getVideoTracks().length, 2,
+       "The inception clone should contain the original video track and a track clone");
+
+    let test = createMediaElement('video', 'testClonePlayback');
+    let playback = new MediaStreamPlayback(test, inceptionClone);
+    await playback.playMedia(false);
+    clones.forEach(c => c.getTracks().forEach(t => t.stop()));
+    stream.getTracks().forEach(t => t.stop());
+  }
+
+  {
+    info("Test adding tracks from many stream clones to the original stream");
+    let stream = gUMStream.clone();
+
+    const LOOPS = 3;
+    for (let i = 0; i < LOOPS; i++) {
+      stream.clone().getTracks().forEach(t => stream.addTrack(t));
+    }
+    is(stream.getAudioTracks().length, Math.pow(2, LOOPS),
+       "The original track should contain the original audio track and all the audio clones");
+    is(stream.getVideoTracks().length, Math.pow(2, LOOPS),
+       "The original track should contain the original video track and all the video clones");
+    stream.getTracks().forEach(t1 => is(stream.getTracks()
+                                              .filter(t2 => t1.id == t2.id)
+                                              .length,
+                                        1, "Each track should be unique"));
+
+    let test = createMediaElement('video', 'testClonePlayback');
+    let playback = new MediaStreamPlayback(test, stream);
+    await playback.playMedia(false);
+  }
+
+  {
+    info("Testing audio content routing with MediaStream.clone()");
+    let ac = new AudioContext();
+
+    let osc1kOriginal = createOscillatorStream(ac, 1000);
+    let audioTrack1kOriginal = osc1kOriginal.getTracks()[0];
+    let audioTrack1kClone = osc1kOriginal.clone().getTracks()[0];
+
+    let osc5kOriginal = createOscillatorStream(ac, 5000);
+    let audioTrack5kOriginal = osc5kOriginal.getTracks()[0];
+    let audioTrack5kClone = osc5kOriginal.clone().getTracks()[0];
+
+    info("Analysing audio output of original stream (1k + 5k)");
+    let stream = new MediaStream();
+    stream.addTrack(audioTrack1kOriginal);
+    stream.addTrack(audioTrack5kOriginal);
+
+    let analyser = new AudioStreamAnalyser(ac, stream);
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            array[analyser.binIndexForFrequency(1000)] > 200 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            array[analyser.binIndexForFrequency(5000)] > 200 &&
+            array[analyser.binIndexForFrequency(10000)] < 50);
+
+    info("Waiting for original tracks to stop");
+    stream.getTracks().forEach(t => t.stop());
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            // WebAudioDestination streams do not handle stop()
+            // XXX Should they? Plan to resolve that in bug 1208384.
+            // array[analyser.binIndexForFrequency(1000)] < 50 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            // array[analyser.binIndexForFrequency(5000)] < 50 &&
+            array[analyser.binIndexForFrequency(10000)] < 50);
+    analyser.disconnect();
+
+    info("Analysing audio output of stream clone (1k + 5k)");
+    stream = new MediaStream();
+    stream.addTrack(audioTrack1kClone);
+    stream.addTrack(audioTrack5kClone);
+
+    analyser = new AudioStreamAnalyser(ac, stream);
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            array[analyser.binIndexForFrequency(1000)] > 200 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            array[analyser.binIndexForFrequency(5000)] > 200 &&
+            array[analyser.binIndexForFrequency(10000)] < 50);
+    analyser.disconnect();
+
+    info("Analysing audio output of clone of clone (1k + 5k)");
+    stream = new MediaStream([audioTrack1kClone, audioTrack5kClone]).clone();
+
+    analyser = new AudioStreamAnalyser(ac, stream);
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            array[analyser.binIndexForFrequency(1000)] > 200 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            array[analyser.binIndexForFrequency(5000)] > 200 &&
+            array[analyser.binIndexForFrequency(10000)] < 50);
+    analyser.disconnect();
+
+    info("Analysing audio output of clone() + addTrack()ed tracks (1k + 5k)");
+    stream = new MediaStream(new MediaStream([ audioTrack1kClone
+                                             , audioTrack5kClone
+                                             ]).clone().getTracks());
+
+    analyser = new AudioStreamAnalyser(ac, stream);
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            array[analyser.binIndexForFrequency(1000)] > 200 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            array[analyser.binIndexForFrequency(5000)] > 200 &&
+            array[analyser.binIndexForFrequency(10000)] < 50);
+    analyser.disconnect();
+
+    info("Analysing audio output of clone()d tracks in original stream (1k) " +
+         "and clone()d tracks in stream clone (5k)");
+    stream = new MediaStream([audioTrack1kClone, audioTrack5kClone]);
+    let streamClone = stream.clone();
+
+    stream.getTracks().forEach(t => stream.removeTrack(t));
+    stream.addTrack(streamClone.getTracks()[0]);
+    streamClone.removeTrack(streamClone.getTracks()[0]);
+
+    analyser = new AudioStreamAnalyser(ac, stream);
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            array[analyser.binIndexForFrequency(1000)] > 200 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            array[analyser.binIndexForFrequency(5000)] < 50);
+    analyser.disconnect();
+
+    let cloneAnalyser = new AudioStreamAnalyser(ac, streamClone);
+    await cloneAnalyser.waitForAnalysisSuccess(array =>
+            array[cloneAnalyser.binIndexForFrequency(1000)] < 50 &&
+            array[cloneAnalyser.binIndexForFrequency(3000)] < 50 &&
+            array[cloneAnalyser.binIndexForFrequency(5000)] > 200 &&
+            array[cloneAnalyser.binIndexForFrequency(10000)] < 50);
+    cloneAnalyser.disconnect();
+
+    info("Analysing audio output enabled and disabled tracks that don't affect each other");
+    stream = new MediaStream([audioTrack1kClone, audioTrack5kClone]);
+    let clone = stream.clone();
+
+    stream.getTracks()[0].enabled = true;
+    stream.getTracks()[1].enabled = false;
+
+    clone.getTracks()[0].enabled = false;
+    clone.getTracks()[1].enabled = true;
+
+    analyser = new AudioStreamAnalyser(ac, stream);
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            array[analyser.binIndexForFrequency(1000)] > 200 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            array[analyser.binIndexForFrequency(5000)] < 50);
+    analyser.disconnect();
+
+    cloneAnalyser = new AudioStreamAnalyser(ac, clone);
+    await cloneAnalyser.waitForAnalysisSuccess(array =>
+            array[cloneAnalyser.binIndexForFrequency(1000)] < 50 &&
+            array[cloneAnalyser.binIndexForFrequency(3000)] < 50 &&
+            array[cloneAnalyser.binIndexForFrequency(5000)] > 200 &&
+            array[cloneAnalyser.binIndexForFrequency(10000)] < 50);
+    cloneAnalyser.disconnect();
+
+    // Restore original tracks
+    stream.getTracks().forEach(t => t.enabled = true);
+  }
+
+  gUMStream.getTracks().forEach(t => t.stop());
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamConstructors.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamConstructors.html
new file mode 100644
index 0000000000..4ea6e3f444
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamConstructors.html
@@ -0,0 +1,171 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  "use strict";
+
+  createHTML({
+    title: "MediaStream constructors with getUserMedia streams Test",
+    bug: "1070216"
+  });
+
+  var audioContext = new AudioContext();
+  var videoElement;
+
+  runTest(() => Promise.resolve()
+    .then(() => videoElement = createMediaElement('video', 'constructorsTest'))
+    .then(() => getUserMedia({video: true})).then(gUMStream => {
+      info("Test default constructor with video");
+      ok(gUMStream.active, "gUMStream with one track should be active");
+      var track = gUMStream.getTracks()[0];
+
+      var stream = new MediaStream();
+      ok(!stream.active, "New MediaStream should be inactive");
+      checkMediaStreamContains(stream, [], "Default constructed stream");
+
+      stream.addTrack(track);
+      ok(stream.active, "MediaStream should be active after adding a track");
+      checkMediaStreamContains(stream, [track], "Added video track");
+
+      var playback = new MediaStreamPlayback(videoElement, stream);
+      return playback.playMedia(false).then(() => {
+        ok(!gUMStream.active, "gUMStream should be inactive after stopping");
+        ok(!stream.active, "stream with stopped tracks should be inactive");
+      });
+    })
+    .then(() => getUserMedia({video: true})).then(gUMStream => {
+      info("Test copy constructor with gUM stream");
+      ok(gUMStream.active, "gUMStream with one track should be active");
+      var track = gUMStream.getTracks()[0];
+
+      var stream = new MediaStream(gUMStream);
+      ok(stream.active, "List constructed MediaStream should be active");
+      checkMediaStreamContains(stream, [track], "Copy constructed video track");
+
+      var playback = new MediaStreamPlayback(videoElement, stream);
+      return playback.playMedia(false).then(() => {
+        ok(!gUMStream.active, "gUMStream should be inactive after stopping");
+        ok(!stream.active, "stream with stopped tracks should be inactive");
+      });
+    })
+    .then(() => getUserMedia({video: true})).then(gUMStream => {
+      info("Test list constructor with empty list");
+      ok(gUMStream.active, "gUMStream with one track should be active");
+      var track = gUMStream.getTracks()[0];
+
+      var stream = new MediaStream([]);
+      ok(!stream.active, "Empty-list constructed MediaStream should be inactive");
+      checkMediaStreamContains(stream, [], "Empty-list constructed stream");
+
+      stream.addTrack(track);
+      ok(stream.active, "MediaStream should be active after adding a track");
+      checkMediaStreamContains(stream, [track], "Added video track");
+
+      var playback = new MediaStreamPlayback(videoElement, stream);
+      return playback.playMedia(false).then(() => {
+        ok(!gUMStream.active, "gUMStream should be inactive after stopping");
+        ok(!stream.active, "stream with stopped tracks should be inactive");
+      });
+    })
+    .then(() => getUserMedia({audio: true, video: true})).then(gUMStream => {
+      info("Test list constructor with a gUM audio/video stream");
+      ok(gUMStream.active, "gUMStream with two tracks should be active");
+      var audioTrack = gUMStream.getAudioTracks()[0];
+      var videoTrack = gUMStream.getVideoTracks()[0];
+
+      var stream = new MediaStream([audioTrack, videoTrack]);
+      ok(stream.active, "List constructed MediaStream should be active");
+      checkMediaStreamContains(stream, [audioTrack, videoTrack],
+                               "List constructed audio and video tracks");
+
+      var playback = new MediaStreamPlayback(videoElement, stream);
+      return playback.playMedia(false).then(() => {
+        ok(!gUMStream.active, "gUMStream should be inactive after stopping");
+        ok(!stream.active, "stream with stopped tracks should be inactive");
+      });
+    })
+    .then(() => getUserMedia({video: true})).then(gUMStream => {
+      info("Test list constructor with gUM-video and WebAudio tracks");
+      ok(gUMStream.active, "gUMStream with one track should be active");
+      var audioStream = createOscillatorStream(audioContext, 2000);
+      ok(audioStream.active, "WebAudio stream should be active");
+
+      var audioTrack = audioStream.getTracks()[0];
+      var videoTrack = gUMStream.getTracks()[0];
+
+      var stream = new MediaStream([audioTrack, videoTrack]);
+      ok(stream.active, "List constructed MediaStream should be active");
+      checkMediaStreamContains(stream, [audioTrack, videoTrack],
+                               "List constructed WebAudio and gUM-video tracks");
+
+      var playback = new MediaStreamPlayback(videoElement, stream);
+      return playback.playMedia(false).then(() => {
+        gUMStream.getTracks().forEach(t => t.stop());
+        ok(!gUMStream.active, "gUMStream should be inactive after stopping");
+        ok(!stream.active, "stream with stopped tracks should be inactive");
+      });
+    })
+    .then(() => {
+      var osc1k = createOscillatorStream(audioContext, 1000);
+      var audioTrack1k = osc1k.getTracks()[0];
+
+      var osc5k = createOscillatorStream(audioContext, 5000);
+      var audioTrack5k = osc5k.getTracks()[0];
+
+      var osc10k = createOscillatorStream(audioContext, 10000);
+      var audioTrack10k = osc10k.getTracks()[0];
+
+      return Promise.resolve().then(() => {
+        info("Analysing audio output with empty default constructed stream");
+        var stream = new MediaStream();
+        var analyser = new AudioStreamAnalyser(audioContext, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+          array[analyser.binIndexForFrequency(1000)]  < 50 &&
+          array[analyser.binIndexForFrequency(5000)]  < 50 &&
+          array[analyser.binIndexForFrequency(10000)] < 50)
+          .then(() => analyser.disconnect());
+      }).then(() => {
+        info("Analysing audio output with copy constructed 5k stream");
+        var stream = new MediaStream(osc5k);
+        is(stream.active, osc5k.active,
+           "Copy constructed MediaStream should preserve active state");
+        var analyser = new AudioStreamAnalyser(audioContext, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+          array[analyser.binIndexForFrequency(1000)]  < 50 &&
+          array[analyser.binIndexForFrequency(5000)]  > 200 &&
+          array[analyser.binIndexForFrequency(10000)] < 50)
+          .then(() => analyser.disconnect());
+      }).then(() => {
+        info("Analysing audio output with empty-list constructed stream");
+        var stream = new MediaStream([]);
+        var analyser = new AudioStreamAnalyser(audioContext, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+          array[analyser.binIndexForFrequency(1000)]  < 50 &&
+          array[analyser.binIndexForFrequency(5000)]  < 50 &&
+          array[analyser.binIndexForFrequency(10000)] < 50)
+          .then(() => analyser.disconnect());
+      }).then(() => {
+        info("Analysing audio output with list constructed 1k, 5k and 10k tracks");
+        var stream = new MediaStream([audioTrack1k, audioTrack5k, audioTrack10k]);
+        ok(stream.active,
+           "List constructed MediaStream from WebAudio should be active");
+        var analyser = new AudioStreamAnalyser(audioContext, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+          array[analyser.binIndexForFrequency(50)]    < 50 &&
+          array[analyser.binIndexForFrequency(1000)]  > 200 &&
+          array[analyser.binIndexForFrequency(2500)]  < 50 &&
+          array[analyser.binIndexForFrequency(5000)]  > 200 &&
+          array[analyser.binIndexForFrequency(7500)]  < 50 &&
+          array[analyser.binIndexForFrequency(10000)] > 200 &&
+          array[analyser.binIndexForFrequency(11000)] < 50)
+          .then(() => analyser.disconnect());
+      });
+    }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamTrackClone.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamTrackClone.html
new file mode 100644
index 0000000000..e5e0764427
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamTrackClone.html
@@ -0,0 +1,170 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  "use strict";
+
+  createHTML({
+    title: "MediaStreamTrack.clone()",
+    bug: "1208371"
+  });
+
+  var testSingleTrackClonePlayback = constraints =>
+      getUserMedia(constraints).then(stream => {
+    info("Test clone()ing an " + constraints + " gUM track");
+    var track = stream.getTracks()[0];
+    var clone = track.clone();
+
+    checkMediaStreamTrackCloneAgainstOriginal(clone, track);
+
+    info("Stopping original track");
+    track.stop();
+
+    info("Creating new stream for clone");
+    var cloneStream = new MediaStream([clone]);
+    checkMediaStreamContains(cloneStream, [clone]);
+
+    info("Testing playback of track clone");
+    var test = createMediaElement('video', 'testClonePlayback');
+    var playback = new MediaStreamPlayback(test, cloneStream);
+    return playback.playMedia(false);
+  });
+
+  runTest(() => Promise.resolve()
+    .then(() => testSingleTrackClonePlayback({audio: true}))
+    .then(() => testSingleTrackClonePlayback({video: true}))
+    .then(() => getUserMedia({video: true})).then(stream => {
+      info("Test cloning a track into inception");
+      var track = stream.getTracks()[0];
+      var clone = track;
+      var clones = Array(10).fill().map(() => clone = clone.clone());
+      var inceptionClone = clones.pop();
+      checkMediaStreamTrackCloneAgainstOriginal(inceptionClone, track);
+
+      var cloneStream = new MediaStream();
+      cloneStream.addTrack(inceptionClone);
+
+      // cloneStream is now essentially the same as stream.clone();
+      checkMediaStreamCloneAgainstOriginal(cloneStream, stream);
+
+      var test = createMediaElement('video', 'testClonePlayback');
+      var playback = new MediaStreamPlayback(test, cloneStream);
+      return playback.playMedia(false).then(() => {
+          info("Testing that clones of ended tracks are ended");
+          cloneStream.clone().getTracks().forEach(t =>
+            is(t.readyState, "ended", "Track " + t.id + " should be ended"));
+        })
+        .then(() => {
+          clones.forEach(t => t.stop());
+          track.stop();
+        });
+    })
+    .then(() => getUserMedia({audio: true, video: true})).then(stream => {
+      info("Test adding many track clones to the original stream");
+
+      const LOOPS = 3;
+      for (var i = 0; i < LOOPS; i++) {
+        stream.getTracks().forEach(t => stream.addTrack(t.clone()));
+      }
+      is(stream.getVideoTracks().length, Math.pow(2, LOOPS),
+         "The original track should contain the original video track and all the video clones");
+      stream.getTracks().forEach(t1 => is(stream.getTracks()
+                                                .filter(t2 => t1.id == t2.id)
+                                                .length,
+                                          1, "Each track should be unique"));
+
+      var test = createMediaElement('video', 'testClonePlayback');
+      var playback = new MediaStreamPlayback(test, stream);
+      return playback.playMedia(false);
+    })
+    .then(() => {
+      info("Testing audio content routing with MediaStreamTrack.clone()");
+      var ac = new AudioContext();
+
+      var osc1kOriginal = createOscillatorStream(ac, 1000);
+      var audioTrack1kOriginal = osc1kOriginal.getTracks()[0];
+      var audioTrack1kClone = audioTrack1kOriginal.clone();
+
+      var osc5kOriginal = createOscillatorStream(ac, 5000);
+      var audioTrack5kOriginal = osc5kOriginal.getTracks()[0];
+      var audioTrack5kClone = audioTrack5kOriginal.clone();
+
+      return Promise.resolve().then(() => {
+        info("Analysing audio output enabled and disabled tracks that don't affect each other");
+        audioTrack1kOriginal.enabled = true;
+        audioTrack5kOriginal.enabled = false;
+
+        audioTrack1kClone.enabled = false;
+        audioTrack5kClone.enabled = true;
+
+        var analyser =
+          new AudioStreamAnalyser(ac, new MediaStream([audioTrack1kOriginal,
+                                                      audioTrack5kOriginal]));
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(50)] < 50 &&
+                 array[analyser.binIndexForFrequency(1000)] > 200 &&
+                 array[analyser.binIndexForFrequency(3000)] < 50 &&
+                 array[analyser.binIndexForFrequency(5000)] < 50)
+          .then(() => analyser.disconnect())
+          .then(() => {
+            var cloneAnalyser =
+              new AudioStreamAnalyser(ac, new MediaStream([audioTrack1kClone,
+                                                           audioTrack5kClone]));
+            return cloneAnalyser.waitForAnalysisSuccess(array =>
+                     array[cloneAnalyser.binIndexForFrequency(1000)] < 50 &&
+                     array[cloneAnalyser.binIndexForFrequency(3000)] < 50 &&
+                     array[cloneAnalyser.binIndexForFrequency(5000)] > 200 &&
+                     array[cloneAnalyser.binIndexForFrequency(10000)] < 50)
+              .then(() => cloneAnalyser.disconnect());
+          })
+          // Restore original tracks
+          .then(() => [audioTrack1kOriginal,
+                       audioTrack5kOriginal,
+                       audioTrack1kClone,
+                       audioTrack5kClone].forEach(t => t.enabled = true));
+      }).then(() => {
+        info("Analysing audio output of 1k original and 5k clone.");
+        var stream = new MediaStream();
+        stream.addTrack(audioTrack1kOriginal);
+        stream.addTrack(audioTrack5kClone);
+
+        var analyser = new AudioStreamAnalyser(ac, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(50)] < 50 &&
+                 array[analyser.binIndexForFrequency(1000)] > 200 &&
+                 array[analyser.binIndexForFrequency(3000)] < 50 &&
+                 array[analyser.binIndexForFrequency(5000)] > 200 &&
+                 array[analyser.binIndexForFrequency(10000)] < 50)
+          .then(() => {
+            info("Waiting for tracks to stop");
+            stream.getTracks().forEach(t => t.stop());
+            return analyser.waitForAnalysisSuccess(array =>
+                     array[analyser.binIndexForFrequency(50)] < 50 &&
+                     array[analyser.binIndexForFrequency(1000)] < 50 &&
+                     array[analyser.binIndexForFrequency(3000)] < 50 &&
+                     array[analyser.binIndexForFrequency(5000)] < 50 &&
+                     array[analyser.binIndexForFrequency(10000)] < 50);
+          }).then(() => analyser.disconnect());
+      }).then(() => {
+        info("Analysing audio output of clones of clones (1kx2 + 5kx4)");
+        var stream = new MediaStream([audioTrack1kClone.clone(),
+                                      audioTrack5kOriginal.clone().clone().clone().clone()]);
+
+        var analyser = new AudioStreamAnalyser(ac, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(50)] < 50 &&
+                 array[analyser.binIndexForFrequency(1000)] > 200 &&
+                 array[analyser.binIndexForFrequency(3000)] < 50 &&
+                 array[analyser.binIndexForFrequency(5000)] > 200 &&
+                 array[analyser.binIndexForFrequency(10000)] < 50)
+          .then(() => analyser.disconnect());
+      });
+    }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_nonDefaultRate.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_nonDefaultRate.html
new file mode 100644
index 0000000000..8a6ac8c62b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_nonDefaultRate.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia feed to a graph with non default rate",
+    bug: "1387454",
+  });
+
+  /**
+   * Run a test to verify that when we use the streem from a gUM to an AudioContext
+   * with non default rate the connection fails. (gUM is always on default rate).
+   */
+  runTest(async () => {
+    // Since we do not examine the stream we do not need loopback.
+    DISABLE_LOOPBACK_TONE = true;
+    const stream = await getUserMedia({audio: true});
+    const nonDefaultRate = 32000;
+    const ac = new AudioContext({sampleRate: nonDefaultRate});
+    mustThrowWith(
+      "Connect stream with graph of different sample rate",
+      "NotSupportedError", () => {
+        ac.createMediaStreamSource(stream);
+      }
+    );
+    for (let t of stream.getTracks()) {
+      t.stop();
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_peerIdentity.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_peerIdentity.html
new file mode 100644
index 0000000000..c4dfb9acb8
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_peerIdentity.html
@@ -0,0 +1,51 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+  <script type="application/javascript" src="blacksilence.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "Test getUserMedia peerIdentity Constraint",
+  bug: "942367"
+});
+async function theTest() {
+  async function testPeerIdentityConstraint(withConstraint) {
+    const config = { audio: true, video: true };
+    if (withConstraint) {
+      config.peerIdentity = 'user@example.com';
+    }
+    info('getting media with constraints: ' + JSON.stringify(config));
+    const stream = await getUserMedia(config);
+    for (const track of stream.getTracks()) {
+      const recorder = new MediaRecorder(new MediaStream([track]));
+      try {
+        recorder.start();
+        ok(!withConstraint,
+          `gUM ${track.kind} track without peerIdentity must not throw`);
+        recorder.stop();
+      } catch(e) {
+        ok(withConstraint,
+          `gUM ${track.kind} track with peerIdentity must throw`);
+      }
+    }
+    await Promise.all([
+      audioIsSilence(withConstraint, stream),
+      videoIsBlack(withConstraint, stream),
+    ]);
+    stream.getTracks().forEach(t => t.stop());
+  };
+
+  // both without and with the constraint
+  await testPeerIdentityConstraint(false);
+  await testPeerIdentityConstraint(true);
+}
+
+runTest(theTest);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission.html
new file mode 100644
index 0000000000..cd02c7326c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission.html
@@ -0,0 +1,104 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({ title: "Test getUserMedia in iframes", bug: "1371741" });
+/**
+  Tests covering enumerateDevices API and deviceId constraint. Exercise code.
+*/
+
+// Call gUM in iframe.
+async function iframeGum(dict, iframe = document.createElement("iframe")) {
+  Object.assign(iframe, dict);
+  if (dict.src) {
+    info(`<iframe src="${dict.src}" sandbox="${dict.sandbox}">`);
+  } else {
+    info(`<iframe srcdoc sandbox="${dict.sandbox}">`);
+  }
+  document.documentElement.appendChild(iframe);
+
+  const once = (t, msg) => new Promise(r => t.addEventListener(msg, r, { once: true }));
+  const haveMessage = once(window, "message");
+  await new Promise(resolve => iframe.onload = resolve);
+  return (await haveMessage).data;
+};
+
+runTest(async () => {
+  const path = "/tests/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission_iframe.html";
+
+  async function sourceFn() {
+    try {
+      const gUM = c => navigator.mediaDevices.getUserMedia(c);
+      let message;
+      let stream;
+      try {
+        stream = await gUM({ video: true });
+        message = 'success';
+      } catch(e) {
+        message = e.name;
+      }
+      parent.postMessage(message, 'https://example.com:443');
+
+      if (message == "success") {
+        stream.getTracks().forEach(track => track.stop());
+      }
+    } catch (e) {
+      setTimeout(() => { throw e; });
+    }
+  }
+
+  const source = `<html\><script\>(${sourceFn.toString()})()</script\></html\>`;
+
+  // Test gUM in sandboxed vs. regular iframe.
+
+  for (const origin of [window.location.origin, "https://test1.example.com"]) {
+    const src = origin + path;
+    is(await iframeGum({ src, sandbox: "allow-scripts" }),
+       "NotAllowedError", "gUM fails in sandboxed iframe " + origin);
+  }
+  is(await iframeGum({
+       src: path,
+       sandbox: "allow-scripts allow-same-origin",
+     }),
+     "success", "gUM works in regular same-origin iframe");
+  is(await iframeGum({
+       src: `https://test1.example.com${path}`,
+       sandbox: "allow-scripts allow-same-origin",
+     }),
+     "NotAllowedError", "gUM fails in regular cross-origin iframe");
+
+  // Test gUM in sandboxed vs regular srcdoc iframe
+
+  const iframeSrcdoc = document.createElement("iframe");
+  iframeSrcdoc.srcdoc = source;
+  is(await iframeGum({ sandbox: "allow-scripts" }, iframeSrcdoc),
+     "NotAllowedError", "gUM fails in sandboxed srcdoc iframe");
+  is(await iframeGum({ sandbox: "allow-scripts allow-same-origin" }, iframeSrcdoc),
+     "success", "gUM works in regular srcdoc iframe");
+
+  // Test gUM in sandboxed vs regular blob iframe
+
+  const blob = new Blob([source], {type : "text/html"});
+	let src = URL.createObjectURL(blob);
+  is(await iframeGum({ src, sandbox: "allow-scripts" }),
+     "NotAllowedError", "gUM fails in sandboxed blob iframe");
+  is(await iframeGum({ src, sandbox: "allow-scripts allow-same-origin"}),
+     "success", "gUM works in regular blob iframe");
+  URL.revokeObjectURL(src);
+
+  // data iframes always have null-principals
+
+  src = `data:text/html;base64,${btoa(source)}`;
+  is(await iframeGum({ src, sandbox: "allow-scripts" }),
+     "NotAllowedError", "gUM fails in sandboxed data iframe");
+  is(await iframeGum({ src, sandbox: "allow-scripts allow-same-origin"}),
+     "NotAllowedError", "gUM fails in regular data iframe");
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission_iframe.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission_iframe.html
new file mode 100644
index 0000000000..732c2cf98c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission_iframe.html
@@ -0,0 +1,30 @@
+<!DOCTYPE HTML>
+<html>
+<body>
+<pre id="test">
+<script type="application/javascript">
+/**
+  Runs inside iframe in test_getUserMedia_permission.html.
+*/
+
+const gUM = c => navigator.mediaDevices.getUserMedia(c);
+
+(async () => {
+  let message;
+  let stream;
+  try {
+    stream = await gUM({ video: true });
+    message = "success";
+  } catch(e) {
+    message = e.name;
+  }
+  parent.postMessage(message, "https://example.com:443");
+
+  if (message == "success") {
+    stream.getTracks().forEach(track => track.stop());
+  }
+})().catch(e => setTimeout(() => { throw e; }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_playAudioTwice.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_playAudioTwice.html
new file mode 100644
index 0000000000..30d168bf38
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_playAudioTwice.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({title: "getUserMedia Play Audio Twice", bug: "822109" });
+  /**
+   * Run a test that we can complete an audio playback cycle twice in a row.
+   */
+  runTest(function () {
+    return getUserMedia({audio: true}).then(audioStream => {
+      var testAudio = createMediaElement('audio', 'testAudio');
+      var playback = new MediaStreamPlayback(testAudio, audioStream);
+
+      return playback.playMediaWithoutStoppingTracks(false)
+        .then(() => playback.playMedia(true));
+    });
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_playVideoAudioTwice.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_playVideoAudioTwice.html
new file mode 100644
index 0000000000..7b5e6effd1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_playVideoAudioTwice.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({title: "getUserMedia Play Video and Audio Twice", bug: "822109" });
+  /**
+   * Run a test that we can complete a video playback cycle twice in a row.
+   */
+  runTest(function () {
+    return getUserMedia({video: true, audio: true}).then(stream => {
+      var testVideo = createMediaElement('video', 'testVideo');
+      var playback = new MediaStreamPlayback(testVideo, stream);
+
+      return playback.playMediaWithoutStoppingTracks(false)
+        .then(() => playback.playMedia(true));
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_playVideoTwice.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_playVideoTwice.html
new file mode 100644
index 0000000000..2890f45eab
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_playVideoTwice.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({ title: "getUserMedia Play Video Twice", bug: "822109" });
+  /**
+   * Run a test that we can complete a video playback cycle twice in a row.
+   */
+  runTest(function () {
+    return getUserMedia({video: true}).then(stream => {
+      var testVideo = createMediaElement('video', 'testVideo');
+      var streamPlayback = new MediaStreamPlayback(testVideo, stream);
+
+      return streamPlayback.playMediaWithoutStoppingTracks(false)
+        .then(() => streamPlayback.playMedia(true));
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_scarySources.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_scarySources.html
new file mode 100644
index 0000000000..782110823e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_scarySources.html
@@ -0,0 +1,51 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+
+createHTML({title: "Detect screensharing sources that are firefox", bug: "1311048"});
+
+const Services = SpecialPowers.Services;
+
+let observe = topic => new Promise(r => Services.obs.addObserver(function o(...args) {
+  Services.obs.removeObserver(o, topic);
+  r(args.map(x => SpecialPowers.wrap(x)));
+}, topic));
+
+let getDevices = async constraints => {
+  SpecialPowers.wrap(document).notifyUserGestureActivation();
+  let [{ windowID, innerWindowID, callID, devices }] = await Promise.race([
+    getUserMedia(constraints),
+    observe("getUserMedia:request")
+  ]);
+  let window = Services.wm.getOuterWindowWithId(windowID);
+  return devices.map(SpecialPowers.wrapCallback(d => d.QueryInterface(Ci.nsIMediaDevice)));
+};
+
+runTest(async () => {
+  await pushPrefs(["media.navigator.permission.disabled", true],
+                  ["media.navigator.permission.force", true]);
+  let devices = await getDevices({video: { mediaSource: "window" }});
+  ok(devices.length, "Found one or more windows.");
+  devices = Array.prototype.filter.call(devices, d => d.scary);
+  ok(devices.length, "Found one or more scary windows (our own counts).");
+  devices = devices.filter(d => d.rawName.includes("MochiTest"));
+  ok(devices.length,
+     "Our own window is among the scary: "
+     + devices.map(d => `"${d.rawName}"`));
+
+  devices = await getDevices({video: { mediaSource: "screen" }});
+  let numScreens = devices.length;
+  ok(numScreens, "Found one or more screens.");
+  devices = Array.prototype.filter.call(devices, d => d.scary);
+  is(devices.length, numScreens, "All screens are scary.");
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_spinEventLoop.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_spinEventLoop.html
new file mode 100644
index 0000000000..ae691785f5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_spinEventLoop.html
@@ -0,0 +1,28 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({ title: "getUserMedia Basic Audio Test", bug: "1208656" });
+  /**
+   * Run a test to verify that we can spin the event loop from within a mozGUM callback.
+   */
+  runTest(() => {
+    var testAudio = createMediaElement('audio', 'testAudio');
+    return new Promise((resolve, reject) => {
+      navigator.mozGetUserMedia({ audio: true }, stream => {
+        SpecialPowers.spinEventLoop(window);
+        ok(true, "Didn't crash");
+        stream.getTracks().forEach(t => t.stop());
+        resolve();
+      }, () => {});
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_trackCloneCleanup.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_trackCloneCleanup.html
new file mode 100644
index 0000000000..60077ec73b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_trackCloneCleanup.html
@@ -0,0 +1,32 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  "use strict";
+
+  createHTML({
+    title: "Stopping a MediaStreamTrack and its clones should deallocate the device",
+    bug: "1294605"
+  });
+
+  runTest(async () => {
+    await pushPrefs(["media.navigator.permission.fake", true]);
+    const stream = await getUserMedia({audio: true, video: true, fake: true});
+    const clone = stream.clone();
+    stream.getTracks().forEach(t => t.stop());
+    stream.clone().getTracks().forEach(t => stream.addTrack(t));
+    is(stream.getTracks().filter(t => t.readyState == "live").length, 0,
+       "Cloning ended tracks should make them ended");
+    [...stream.getTracks(), ...clone.getTracks()].forEach(t => t.stop());
+
+    // Bug 1295352: better to be explicit about noGum here wrt future refactoring.
+    return noGum();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_trackEnded.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_trackEnded.html
new file mode 100644
index 0000000000..b275f4555f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_trackEnded.html
@@ -0,0 +1,68 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<iframe id="iframe" onload="iframeLoaded()" srcdoc="
+  <script type='application/javascript'>
+  document.gUM = (constraints, success, failure) =>
+    navigator.mediaDevices.getUserMedia(constraints).then(success, failure);
+  </script>">
+</iframe>
+<script type="application/javascript">
+  "use strict";
+  let iframeLoadedPromise = {};
+  iframeLoadedPromise.promise = new Promise(r => {
+    iframeLoadedPromise.resolve = r;
+  });;
+  function iframeLoaded() {
+    iframeLoadedPromise.resolve();
+  }
+
+  createHTML({
+    title: "getUserMedia MediaStreamTrack 'ended' event on navigating",
+    bug: "1208373",
+  });
+
+  runTest(async () => {
+    await iframeLoadedPromise.promise;
+    let iframe = document.getElementById("iframe");
+    let stream;
+    // We're passing callbacks into a method in the iframe here, because
+    // a Promise created in the iframe is unusable after the iframe has
+    // navigated away (see bug 1269400 for details).
+    return new Promise((resolve, reject) =>
+        iframe.contentDocument.gUM({audio: true, video: true}, resolve, reject))
+      .then(s => {
+        // We're cloning a stream containing identical tracks (an original
+        // and its clone) to test that ended works both for originals
+        // clones when they're both owned by the same MediaStream.
+        // (Bug 1274221)
+        stream = new MediaStream([].concat(s.getTracks(), s.getTracks())
+                                   .map(t => t.clone())).clone();
+        var allTracksEnded = Promise.all(stream.getTracks().map(t => {
+          info("Set up ended handler for track " + t.id);
+          return haveEvent(t, "ended", wait(50000))
+            .then(event => {
+              info("ended handler invoked for track " + t.id);
+              is(event.target, t, "Target should be correct");
+            }, e => e ? Promise.reject(e)
+                      : ok(false, "ended event never raised for track " + t.id));
+        }));
+        stream.getTracks().forEach(t =>
+          is(t.readyState, "live",
+             "Non-ended track should have readyState 'live'"));
+        iframe.srcdoc = "";
+        info("iframe has been reset. Waiting for tracks to end.");
+        return allTracksEnded;
+      })
+      .then(() => stream.getTracks().forEach(t =>
+        is(t.readyState, "ended",
+           "Ended track should have readyState 'ended'")));
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_groupId.html b/dom/media/webrtc/tests/mochitests/test_groupId.html
new file mode 100644
index 0000000000..f2aefe5e80
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_groupId.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({ title: "Test group id of MediaDeviceInfo", bug: "1213453" });
+
+async function getDefaultDevices() {
+  const devices = await navigator.mediaDevices.enumerateDevices();
+  is(devices.length, 2, "Two fake devices found.");
+
+  devices.forEach(d => isnot(d.groupId, "", "GroupId is included in every device"));
+
+  const videos = devices.filter(d => d.kind == "videoinput");
+  is(videos.length, 1, "One video device found.");
+  const audios = devices.filter(d => d.kind == "audioinput");
+  is(audios.length, 1, "One microphone device found.");
+
+  return {audio: audios[0], video: videos[0]};
+}
+
+runTest(async () => {
+  // Force fake devices in order to be able to change camera name by pref.
+  await pushPrefs(["media.navigator.streams.fake", true],
+                  ["media.audio_loopback_dev", ""],
+                  ["media.video_loopback_dev", ""]);
+
+  const afterGum = await navigator.mediaDevices.getUserMedia({
+    video: true, audio: true
+  });
+  afterGum.getTracks().forEach(track => track.stop());
+
+  let {audio, video} = await getDefaultDevices();
+
+  /* The low level method to correlate groupIds is by device names.
+   * Use a similar comparison here to verify that it works.
+   * Multiple devices of the same device name are not expected in
+   * automation. */
+  isnot(audio.label, video.label, "Audio label differs from video");
+  isnot(audio.groupId, video.groupId, "Not the same groupIds");
+  // Change video name to match.
+  await pushPrefs(["media.getusermedia.fake-camera-name", audio.label]);
+  ({audio, video} = await getDefaultDevices());
+  is(audio.label, video.label, "Audio label matches video");
+  is(audio.groupId, video.groupId, "GroupIds should be the same");
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_multi_mics.html b/dom/media/webrtc/tests/mochitests/test_multi_mics.html
new file mode 100644
index 0000000000..95bcbfd3e4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_multi_mics.html
@@ -0,0 +1,66 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "Test the ability of opening multiple microphones via gUM",
+  bug: "1238038",
+});
+
+runTest(async () => {
+  // Ensure we use the real microphones by disabling loopback devices and fake devices.
+  await pushPrefs(["media.audio_loopback_dev", ""], ["media.navigator.streams.fake", false]);
+
+  try {
+    let devices = await navigator.mediaDevices.enumerateDevices();
+    // Create constraints
+    let constraints = [];
+    devices.forEach((device) => {
+      if (device.kind === "audioinput") {
+        constraints.push({
+          audio: { deviceId: { exact: device.deviceId } },
+        });
+      }
+    });
+    if (constraints.length >= 2) {
+      // Create constraints
+      let constraints = [];
+      devices.forEach((device) => {
+        if (device.kind === "audioinput") {
+          constraints.push({
+            audio: { deviceId: { exact: device.deviceId } },
+          });
+        }
+      });
+      // Open microphones by the constraints
+      let mediaStreams = [];
+      for (let c of constraints) {
+        let stream = await navigator.mediaDevices.getUserMedia(c);
+        dump("MediaStream: " + stream.id + " for device: " + c.audio.deviceId.exact + " is created\n");
+        mediaStreams.push(stream);
+      }
+      // Close microphones
+      for (let stream of mediaStreams) {
+        for (let track of stream.getTracks()) {
+          track.stop();
+        }
+        dump("Stop all tracks in MediaStream: " + stream.id + "\n");
+      }
+      mediaStreams = [];
+    } else {
+      dump("Skip test since we need at least two microphones\n");
+    }
+  } catch (e) {
+    ok(false, e.name + ": " + e.message);
+  }
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_ondevicechange.html b/dom/media/webrtc/tests/mochitests/test_ondevicechange.html
new file mode 100644
index 0000000000..4358d9d748
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_ondevicechange.html
@@ -0,0 +1,180 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <meta charset="utf-8">
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "ondevicechange tests",
+  bug: "1152383"
+});
+
+async function resolveOnEvent(target, name) {
+  return new Promise(r => target.addEventListener(name, r, {once: true}));
+}
+let eventCount = 0;
+async function triggerVideoDevicechange() {
+  ++eventCount;
+  // "media.getusermedia.fake-camera-name" specifies the name of the single
+  // fake video camera.
+  // Changing the pref imitates replacing one device with another.
+  return pushPrefs(["media.getusermedia.fake-camera-name",
+                    `devicechange ${eventCount}`])
+}
+function addIframe() {
+  const iframe = document.createElement("iframe");
+  // Workaround for bug 1743933
+  iframe.loadPromise = resolveOnEvent(iframe, "load");
+  document.documentElement.appendChild(iframe);
+  return iframe;
+}
+
+runTest(async () => {
+  // A toplevel Window and an iframe Windows are compared for devicechange
+  // events.
+  const iframe1 = addIframe();
+  const iframe2 = addIframe();
+  await Promise.all([
+    iframe1.loadPromise,
+    iframe2.loadPromise,
+    pushPrefs(
+      // Use the fake video backend to trigger devicechange events.
+      ["media.navigator.streams.fake", true],
+      // Loopback would override fake.
+      ["media.video_loopback_dev", ""],
+      // Make fake devices count as real, permission-wise, or devicechange
+      // events won't be exposed
+      ["media.navigator.permission.fake", true],
+      // For gUM.
+      ["media.navigator.permission.disabled", true]
+    ),
+  ]);
+  const topDevices = navigator.mediaDevices;
+  const frame1Devices = iframe1.contentWindow.navigator.mediaDevices;
+  const frame2Devices = iframe2.contentWindow.navigator.mediaDevices;
+  // Initialization of MediaDevices::mLastPhysicalDevices is triggered when
+  // ondevicechange is set but tests "media.getusermedia.fake-camera-name"
+  // asynchronously.  Wait for getUserMedia() completion to ensure that the
+  // pref has been read before doDevicechanges() changes it.
+  frame1Devices.ondevicechange = () => {};
+  const topEventPromise = resolveOnEvent(topDevices, "devicechange");
+  const frame2EventPromise = resolveOnEvent(frame2Devices, "devicechange");
+  (await frame1Devices.getUserMedia({video: true})).getTracks()[0].stop();
+
+  await Promise.all([
+    resolveOnEvent(frame1Devices, "devicechange"),
+    triggerVideoDevicechange(),
+  ]);
+  ok(true,
+     "devicechange event is fired when gUM has been in use");
+  // The number of devices has not changed.  Race a settled Promise to check
+  // that no devicechange event has been received in frame2.
+  const racer = {};
+  is(await Promise.race([frame2EventPromise, racer]), racer,
+     "devicechange event is NOT fired in iframe2 for replaced device when " +
+     "gUM has NOT been in use");
+  // getUserMedia() is invoked on frame2Devices after a first device list
+  // change but before returning to the previous state, in order to test that
+  // the device set is compared with the set after previous device list
+  // changes regardless of whether a "devicechange" event was previously
+  // dispatched.
+  (await frame2Devices.getUserMedia({video: true})).getTracks()[0].stop();
+  // Revert device list change.
+  await Promise.all([
+    resolveOnEvent(frame1Devices, "devicechange"),
+    resolveOnEvent(frame2Devices, "devicechange"),
+    SpecialPowers.popPrefEnv(),
+  ]);
+  ok(true,
+     "devicechange event is fired on return to previous list " +
+     "after gUM has been is use");
+
+  const frame1EventPromise1 = resolveOnEvent(frame1Devices, "devicechange");
+  while (true) {
+    const racePromise = Promise.race([
+      frame1EventPromise1,
+      // 100ms is half the coalescing time in MediaManager::DeviceListChanged().
+      wait(100, {type: "wait done"}),
+    ]);
+    await triggerVideoDevicechange();
+    if ((await racePromise).type == "devicechange") {
+      ok(true,
+         "devicechange event is fired even when hardware changes continue");
+      break;
+    }
+  }
+
+  is(await Promise.race([topEventPromise, racer]), racer,
+     "devicechange event is NOT fired for device replacements when " +
+     "gUM has NOT been in use");
+
+  if (navigator.userAgent.includes("Android")) {
+    todo(false, "test assumes Firefox-for-Desktop specific API and behavior");
+    return;
+  }
+  // Open a new tab, which is expected to receive focus and hide the first tab.
+  const tab = window.open();
+  SimpleTest.registerCleanupFunction(() => tab.close());
+  await Promise.all([
+    resolveOnEvent(document, 'visibilitychange'),
+    resolveOnEvent(tab, 'focus'),
+  ]);
+  ok(tab.document.hasFocus(), "tab.document.hasFocus()");
+  await Promise.all([
+    resolveOnEvent(tab, 'blur'),
+    SpecialPowers.spawnChrome([], function focusUrlBar() {
+      this.browsingContext.topChromeWindow.gURLBar.focus();
+    }),
+  ]);
+  ok(!tab.document.hasFocus(), "!tab.document.hasFocus()");
+  is(document.visibilityState, 'hidden', 'visibilityState')
+  const frame1EventPromise2 = resolveOnEvent(frame1Devices, "devicechange");
+  const tabDevices = tab.navigator.mediaDevices;
+  tabDevices.ondevicechange = () => {};
+  const tabStream = await tabDevices.getUserMedia({video: true});
+  // Trigger and await two devicechanges on tabDevices to wait long enough to
+  // provide that a devicechange on another MediaDevices would be received.
+  for (let i = 0; i < 2; ++i) {
+    await Promise.all([
+      resolveOnEvent(tabDevices, "devicechange"),
+      triggerVideoDevicechange(),
+    ]);
+  };
+  is(await Promise.race([frame1EventPromise2, racer]), racer,
+     "devicechange event is NOT fired while tab is in background");
+  tab.close();
+  await resolveOnEvent(document, 'visibilitychange');
+  is(document.visibilityState, 'visible', 'visibilityState')
+  await frame1EventPromise2;
+  ok(true, "devicechange event IS fired when tab returns to foreground");
+
+  const audioLoopbackDev =
+        SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (!navigator.userAgent.includes("Linux")) {
+    todo_isnot(audioLoopbackDev, "", "audio_loopback_dev");
+    return;
+  }
+  isnot(audioLoopbackDev, "", "audio_loopback_dev");
+  await Promise.all([
+    resolveOnEvent(topDevices, "devicechange"),
+    pushPrefs(["media.audio_loopback_dev", "none"]),
+  ]);
+  ok(true,
+     "devicechange event IS fired when last audio device is removed and " +
+     "gUM has NOT been in use");
+  await Promise.all([
+    resolveOnEvent(topDevices, "devicechange"),
+    pushPrefs(["media.audio_loopback_dev", audioLoopbackDev]),
+  ]);
+  ok(true,
+     "devicechange event IS fired when first audio device is added and " +
+     "gUM has NOT been in use");
+});
+
+</script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addAudioTrackToExistingVideoStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addAudioTrackToExistingVideoStream.html
new file mode 100644
index 0000000000..b09d7ffeb5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addAudioTrackToExistingVideoStream.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1246310",
+    title: "Renegotiation: add audio track to existing video-only stream",
+  });
+
+  runNetworkTest(function (options) {
+    SimpleTest.requestCompleteLog();
+    const test = new PeerConnectionTest(options);
+    test.chain.replace("PC_LOCAL_GUM",
+      [
+        function PC_LOCAL_GUM_ATTACH_VIDEO_ONLY(test) {
+          var localConstraints = {audio: true, video: true};
+          test.setMediaConstraints([{video: true}], []);
+          return getUserMedia(localConstraints)
+            .then(s => test.originalGumStream = s)
+            .then(() => is(test.originalGumStream.getAudioTracks().length, 1,
+                           "Should have 1 audio track"))
+            .then(() => is(test.originalGumStream.getVideoTracks().length, 1,
+                           "Should have 1 video track"))
+            .then(() => test.pcLocal.attachLocalTrack(
+                          test.originalGumStream.getVideoTracks()[0],
+                          test.originalGumStream));
+        },
+      ]
+    );
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ATTACH_SECOND_TRACK_AUDIO(test) {
+          test.setMediaConstraints([{audio: true, video: true}], []);
+          return test.pcLocal.attachLocalTrack(
+                   test.originalGumStream.getAudioTracks()[0],
+                   test.originalGumStream);
+        },
+      ],
+      [
+        function PC_CHECK_REMOTE_AUDIO_FLOW(test) {
+          return test.pcRemote.checkReceivingToneFrom(new AudioContext(), test.pcLocal);
+        }
+      ]
+    );
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addDataChannel.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addDataChannel.html
new file mode 100644
index 0000000000..c7536214e5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addDataChannel.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: add DataChannel"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+                     commandsCreateDataChannel,
+                     commandsCheckDataChannel);
+
+    // Insert before the second PC_LOCAL_WAIT_FOR_MEDIA_FLOW
+    test.chain.insertBefore('PC_LOCAL_WAIT_FOR_MEDIA_FLOW',
+                            commandsWaitForDataChannel,
+                            false,
+                            1);
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addDataChannelNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addDataChannelNoBundle.html
new file mode 100644
index 0000000000..6ad754336c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addDataChannelNoBundle.html
@@ -0,0 +1,44 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: add DataChannel"
+  });
+
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.bundle = false;
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+                     commandsCreateDataChannel.concat(
+                       [
+                         function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+                           test.pcLocal.expectIceChecking();
+                         },
+                         function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+                           test.pcRemote.expectIceChecking();
+                         },
+                       ]
+                      ),
+                     commandsCheckDataChannel);
+
+    // Insert before the second PC_LOCAL_WAIT_FOR_MEDIA_FLOW
+    test.chain.insertBefore('PC_LOCAL_WAIT_FOR_MEDIA_FLOW',
+                            commandsWaitForDataChannel,
+                            false,
+                            1);
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondAudioStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondAudioStream.html
new file mode 100644
index 0000000000..61a0250887
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondAudioStream.html
@@ -0,0 +1,45 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: add second audio stream"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}, {audio: true}],
+                                   [{audio: true}]);
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_ADDED_TRACK(test) {
+          // We test both tracks to avoid an ordering problem
+          is(test.pcRemote._pc.getReceivers().length, 2,
+             "pcRemote should have two receivers");
+          return Promise.all(test.pcRemote._pc.getReceivers().map(r => {
+            const analyser = new AudioStreamAnalyser(
+                new AudioContext(), new MediaStream([r.track]));
+            const freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+            return analyser.waitForAnalysisSuccess(arr => arr[freq] > 200);
+          }));
+        },
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondAudioStreamNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondAudioStreamNoBundle.html
new file mode 100644
index 0000000000..32d0564717
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondAudioStreamNoBundle.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: add second audio stream, no bundle"
+  });
+
+  runNetworkTest(function (options = {}) {
+    options.bundle = false;
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}, {audio: true}],
+                                   [{audio: true}]);
+          // Since this is a NoBundle variant, adding a track will cause us to
+          // go back to checking.
+          test.pcLocal.expectIceChecking();
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_ADDED_TRACK(test) {
+          // We test both tracks to avoid an ordering problem
+          is(test.pcRemote._pc.getReceivers().length, 2,
+             "pcRemote should have two receivers");
+          return Promise.all(test.pcRemote._pc.getReceivers().map(r => {
+            const analyser = new AudioStreamAnalyser(
+                new AudioContext(), new MediaStream([r.track]));
+            const freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+            return analyser.waitForAnalysisSuccess(arr => arr[freq] > 200);
+          }));
+        },
+      ]
+    );
+
+    // TODO(bug 1093835): figure out how to verify if media flows through the new stream
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondVideoStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondVideoStream.html
new file mode 100644
index 0000000000..1565958d01
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondVideoStream.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: add second video stream"
+  });
+
+  runNetworkTest(async function (options) {
+    // Use fake video here since the native fake device on linux doesn't
+    // change color as needed by checkVideoPlaying() below.
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{video: true}, {video: true}],
+                                   [{video: true}]);
+          return test.pcLocal.getAllUserMediaAndAddStreams([{video: true}]);
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_VIDEO_FLOW(test) {
+          const h = new VideoStreamHelper();
+          is(test.pcRemote.remoteMediaElements.length, 2,
+             "Should have two remote media elements after renegotiation");
+          return Promise.all(test.pcRemote.remoteMediaElements.map(video =>
+            h.checkVideoPlaying(video)));
+        },
+      ]
+    );
+
+    test.setMediaConstraints([{video: true, fake: true}], [{video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondVideoStreamNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondVideoStreamNoBundle.html
new file mode 100644
index 0000000000..2857100998
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondVideoStreamNoBundle.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: add second video stream, no bundle"
+  });
+
+  runNetworkTest(async function (options = {}) {
+    // Use fake video here since the native fake device on linux doesn't
+    // change color as needed by checkVideoPlaying() below.
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    options.bundle = false;
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{video: true}, {video: true}],
+                                   [{video: true}]);
+          // Since this is a NoBundle variant, adding a track will cause us to
+          // go back to checking.
+          test.pcLocal.expectIceChecking();
+          return test.pcLocal.getAllUserMediaAndAddStreams([{video: true}]);
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_VIDEO_FLOW(test) {
+          const h = new VideoStreamHelper();
+          is(test.pcRemote.remoteMediaElements.length, 2,
+             "Should have two remote media elements after renegotiation");
+          return Promise.all(test.pcRemote.remoteMediaElements.map(video =>
+            h.checkVideoPlaying(video)));
+        },
+      ]
+    );
+
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addtrack_removetrack_events.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addtrack_removetrack_events.html
new file mode 100644
index 0000000000..ff9ca9a772
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addtrack_removetrack_events.html
@@ -0,0 +1,75 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "MediaStream's 'addtrack' and 'removetrack' events with gUM",
+  bug: "1208328"
+});
+
+runNetworkTest(function (options) {
+  let test = new PeerConnectionTest(options);
+  let eventsPromise;
+  addRenegotiation(test.chain,
+    [
+      function PC_LOCAL_SWAP_VIDEO_TRACKS(test) {
+        return getUserMedia({video: true}).then(stream => {
+          var videoTransceiver = test.pcLocal._pc.getTransceivers()[1];
+          is(videoTransceiver.currentDirection, "sendonly",
+             "Video transceiver's current direction is sendonly");
+          is(videoTransceiver.direction, "sendrecv",
+             "Video transceiver's desired direction is sendrecv");
+
+          const localStream = test.pcLocal._pc.getLocalStreams()[0];
+          ok(localStream, "Should have local stream");
+
+          const remoteStream = test.pcRemote._pc.getRemoteStreams()[0];
+          ok(remoteStream, "Should have remote stream");
+
+          const newTrack = stream.getTracks()[0];
+
+          const videoSenderIndex =
+            test.pcLocal._pc.getSenders().findIndex(s => s.track.kind == "video");
+          isnot(videoSenderIndex, -1, "Should have video sender");
+
+          test.pcLocal.removeSender(videoSenderIndex);
+          is(videoTransceiver.direction, "recvonly",
+             "Video transceiver should be recvonly after removeTrack");
+          test.pcLocal.attachLocalTrack(stream.getTracks()[0], localStream);
+          is(videoTransceiver.direction, "recvonly",
+             "Video transceiver should be recvonly after addTrack");
+
+          eventsPromise = haveEvent(remoteStream, "addtrack",
+              wait(50000, new Error("No addtrack event for " + newTrack.id)))
+            .then(trackEvent => {
+              ok(trackEvent instanceof MediaStreamTrackEvent,
+                 "Expected event to be instance of MediaStreamTrackEvent");
+              is(trackEvent.type, "addtrack",
+                 "Expected addtrack event type");
+              is(trackEvent.track.readyState, "live",
+                 "added track should be live");
+            })
+            .then(() => haveNoEvent(remoteStream, "addtrack"));
+        });
+      },
+    ],
+    [
+      function PC_REMOTE_CHECK_EVENTS(test) {
+        return eventsPromise;
+      },
+    ]
+  );
+
+  test.setMediaConstraints([{audio: true, video: true}], []);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_answererAddSecondAudioStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_answererAddSecondAudioStream.html
new file mode 100644
index 0000000000..d9b01bf722
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_answererAddSecondAudioStream.html
@@ -0,0 +1,32 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: answerer adds second audio stream"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    addRenegotiationAnswerer(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}, {audio: true}],
+                                   [{audio: true}]);
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioChannels.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioChannels.html
new file mode 100644
index 0000000000..f6e77f8271
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioChannels.html
@@ -0,0 +1,102 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+
+createHTML({
+  bug: "1765005",
+  title: "Verify audio channel limits for each negotiated audio codec",
+});
+
+const matchesChannels = (sdp, codec, channels) =>
+  !!sdp.match(new RegExp(`a=rtpmap:\\d* ${codec}\/\\d*\/${channels}\r\n`, "g")) ||
+  (channels <= 1 &&
+   !!sdp.match(new RegExp(`a=rtpmap:\\d* ${codec}\/\\d*\r\n`, "g")));
+
+async function testAudioChannels(track, codec, channels, accepted, expectedChannels) {
+  const pc1 = new RTCPeerConnection();
+  const pc2 = new RTCPeerConnection();
+  try {
+    pc1.addTrack(track);
+    pc1.onicecandidate = e => pc2.addIceCandidate(e.candidate);
+    pc2.onicecandidate = e => pc1.addIceCandidate(e.candidate);
+    await pc1.setLocalDescription();
+    await pc2.setRemoteDescription(pc1.localDescription);
+    let {type, sdp} = await pc2.createAnswer();
+    sdp = sdp.replace(new RegExp(`a=rtpmap:(\\d*) ${codec}\/(\\d*)\/?\\d*\r\n`, "g"),
+                                 `a=rtpmap:$1 ${codec}/$2/${channels}\r\n`);
+    const payloadType = Number(sdputils.findCodecId(sdp, codec));
+    sdp = sdputils.removeAllButPayloadType(sdp, payloadType);
+    ok(matchesChannels(sdp, codec, channels), "control");
+    await pc2.setLocalDescription({type, sdp});
+    is(matchesChannels(pc2.localDescription.sdp, codec, channels),
+       accepted,
+       `test pc2.localDescription`);
+    try {
+      await pc1.setRemoteDescription(pc2.localDescription);
+      ok(expectedChannels, "SRD should succeed iff we're expecting channels");
+      const [receiver] = pc2.getReceivers();
+      await new Promise(r => receiver.track.onunmute = r);
+      let stats = await receiver.getStats();
+      let inboundStat = [...stats.values()].find(({type}) => type == "inbound-rtp");
+      if (!inboundStat) {
+        info("work around bug 1765215"); // TODO bug 1765215
+        await new Promise(r => setTimeout(r, 200));
+        stats = await receiver.getStats();
+        inboundStat = [...stats.values()].find(({type}) => type == "inbound-rtp");
+      }
+      ok(inboundStat, "has inbound-rtp stats after track unmute (w/workaround)");
+      const codecStat = stats.get(inboundStat.codecId);
+      ok(codecStat.mimeType.includes(codec), "correct codec");
+      is(codecStat.payloadType, payloadType, "correct payloadType");
+      is(codecStat.channels, expectedChannels, "expected channels");
+    } catch (e) {
+      ok(!expectedChannels, "SRD should fail iff we're not expecting channels");
+    }
+  } finally {
+    pc1.close();
+    pc2.close();
+  }
+}
+
+runNetworkTest(async () => {
+  const [track] = (await navigator.mediaDevices.getUserMedia({audio: true}))
+      .getAudioTracks();
+  try {
+    for (let [codec, channels, accepted, expectedChannels] of [
+      ["opus", 2, true, 2],
+      ["opus", 1, true, 0],
+      ["opus", 1000, true, 0],
+      ["G722", 1, true, 1],
+      ["G722", 2, true, 0],
+      ["G722", 1000, true, 0],
+      ["PCMU", 1, true, 1],
+      ["PCMU", 2, false, 1],
+      ["PCMU", 1000, false, 1],
+      ["PCMA", 1, true, 1],
+      ["PCMA", 2, false, 1],
+      ["PCMA", 1000, false, 1]
+    ]) {
+      const testName = `${codec} with ${channels} channel(s) is ` +
+          `${accepted? "accepted" : "ignored"} and produces ` +
+          `${expectedChannels || "no"} channels`;
+      try {
+        info(`Testing that ${testName}`);
+        await testAudioChannels(track, codec, channels, accepted, expectedChannels);
+      } catch (e) {
+        ok(false, `Error testing that ${testName}: ${e}\n${e.stack}`);
+      }
+    }
+  } finally {
+    track.stop();
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioCodecs.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioCodecs.html
new file mode 100644
index 0000000000..8874436e3b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioCodecs.html
@@ -0,0 +1,81 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1395853",
+    title: "Verify audio content over WebRTC for every audio codec",
+  });
+
+  // We match the format member against the sdp to figure out the payload type,
+  // So all other present codecs can be removed.
+  const codecs = [ "opus", "G722", "PCMU", "PCMA" ];
+
+  async function testAudioCodec(options = {}, codec) {
+    // sdputils checks for opus as part of its sdp sanity test
+    options.opus = codec == "opus";
+
+    let test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], []);
+
+    test.chain.insertBefore("PC_LOCAL_SET_LOCAL_DESCRIPTION", [
+      function PC_LOCAL_FILTER_OUT_CODECS() {
+        let otherCodec = codecs.find(c => c != codec);
+        let otherId = sdputils.findCodecId(test.originalOffer.sdp, otherCodec);
+
+        let id = sdputils.findCodecId(test.originalOffer.sdp, codec);
+        test.originalOffer.sdp =
+          sdputils.removeAllButPayloadType(test.originalOffer.sdp, id);
+
+        ok(!test.originalOffer.sdp.match(new RegExp(`m=.*UDP/TLS/RTP/SAVPF.* ${otherId}[^0-9]`, "gi")),
+          `Other codec ${otherId} should be removed after filtering`);
+        ok(test.originalOffer.sdp.match(new RegExp(`m=.*UDP/TLS/RTP/SAVPF.* ${id}[^0-9]`, "gi")),
+          `Tested codec ${id} should remain after filtering`);
+
+        for (let c of codecs.filter(c => c != codec)) {
+          // Remove rtpmaps for the other codecs so sdp sanity tests pass.
+          let id = sdputils.findCodecId(test.originalOffer.sdp, c);
+          test.originalOffer.sdp =
+            sdputils.removeRtpMapForPayloadType(test.originalOffer.sdp, id);
+        }
+
+        ok(!test.originalOffer.sdp.match(new RegExp(`a=rtpmap:${otherId}.*\\r\\n`, "gi")),
+           `Rtpmap of other codec ${otherId} should be removed after filtering`);
+        ok(test.originalOffer.sdp.match(new RegExp(`a=rtpmap:${id}.*\\r\\n`, "gi")),
+           `Rtpmap of tested codec should remain after filtering`);
+      },
+    ]);
+
+    test.chain.append([
+      async function CHECK_AUDIO_FLOW() {
+        try {
+          await test.pcRemote.checkReceivingToneFrom(new AudioContext(), test.pcLocal);
+          ok(true, "input and output audio data matches");
+        } catch(e) {
+          ok(false, `No audio flow: ${e}`);
+        }
+      },
+    ]);
+
+    await test.run();
+  }
+
+  runNetworkTest(async (options) => {
+    for (let codec of codecs) {
+      info(`Testing audio for codec ${codec}`);
+      try {
+        await testAudioCodec(options, codec);
+      } catch(e) {
+        ok(false, `Error in test for codec ${codec}: ${e}\n${e.stack}`);
+      }
+      info(`Tested audio for codec ${codec}`);
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioContributingSources.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioContributingSources.html
new file mode 100644
index 0000000000..333b40a888
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioContributingSources.html
@@ -0,0 +1,144 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1363667",
+    title: "Test audio receiver getContributingSources"
+  });
+
+  // test_peerConnection_audioSynchronizationSources.html tests
+  // much of the functionality of getContributingSources as the implementation
+  // is shared.
+  var testGetContributingSources = async (test) => {
+    const remoteReceiver = test.pcRemote.getReceivers()[0];
+    const localReceiver = test.pcLocal.getReceivers()[0];
+
+    // Check that getContributingSources is empty as there is no MCU
+    is(remoteReceiver.getContributingSources().length, 0,
+       "remote contributing sources is empty");
+    is(localReceiver.getContributingSources().length, 0,
+       "local contributing sources is empty");
+    // Wait for the next JS event loop iteration, to clear the cache
+    await Promise.resolve().then();
+    // Insert new entries as if there were an MCU
+    const csrc0 = 124756;
+    const timestamp0 = performance.now() + performance.timeOrigin;
+    const rtpTimestamp0 = 11111;
+    const hasAudioLevel0 = true;
+    // Audio level as expected to be received in RTP
+    const audioLevel0 = 34;
+    // Audio level as expected to be returned
+    const expectedAudioLevel0 = 10 ** (-audioLevel0 / 20);
+
+    SpecialPowers.wrap(remoteReceiver).mozInsertAudioLevelForContributingSource(
+        csrc0,
+        timestamp0,
+        rtpTimestamp0,
+        hasAudioLevel0,
+        audioLevel0);
+
+    const csrc1 = 5786;
+    const timestamp1 = timestamp0 - 200;
+    const rtpTimestamp1 = 22222;
+    const hasAudioLevel1 = false;
+    const audioLevel1 = 0;
+
+    SpecialPowers.wrap(remoteReceiver).mozInsertAudioLevelForContributingSource(
+        csrc1,
+        timestamp1,
+        rtpTimestamp1,
+        hasAudioLevel1,
+        audioLevel1);
+
+    const csrc2 = 93487;
+    const timestamp2 = timestamp0 - 200;
+    const rtpTimestamp2 = 333333;
+    const hasAudioLevel2 = true;
+    const audioLevel2 = 127;
+
+    SpecialPowers.wrap(remoteReceiver).mozInsertAudioLevelForContributingSource(
+        csrc2,
+        timestamp2,
+        rtpTimestamp2,
+        hasAudioLevel2,
+        audioLevel2);
+
+    const contributingSources = remoteReceiver.getContributingSources();
+    is(contributingSources.length, 3,
+       "Expected number of contributing sources");
+
+    // Check that both inserted were returned
+    const source0 = contributingSources.find(c => c.source == csrc0);
+    ok(source0, "first csrc was found");
+
+    const source1 = contributingSources.find(c => c.source == csrc1);
+    ok(source1, "second csrsc was found");
+
+    // Add a small margin of error in the timestamps
+    const compareTimestamps = (ts1, ts2) => Math.abs(ts1 - ts2) < 100;
+
+    // Check the CSRC with audioLevel
+    const isWithinErr = Math.abs(source0.audioLevel - expectedAudioLevel0)
+        < expectedAudioLevel0 / 50;
+    ok(isWithinErr,
+       `Contributing source has correct audio level. (${source0.audioLevel})`);
+    ok(compareTimestamps(source0.timestamp, timestamp0),
+        `Contributing source has correct timestamp (got ${source0.timestamp}), expected ${timestamp0}`);
+    is(source0.rtpTimestamp, rtpTimestamp0,
+       `Contributing source has correct RTP timestamp (${source0.rtpTimestamp}`);
+    // Check the CSRC without audioLevel
+    is(source1.audioLevel, undefined,
+       `Contributing source has no audio level. (${source1.audioLevel})`);
+    ok(compareTimestamps(source1.timestamp, timestamp1),
+        `Contributing source has correct timestamp (got ${source1.timestamp}, expected ${timestamp1})`);
+    is(source1.rtpTimestamp, rtpTimestamp1,
+       `Contributing source has correct RTP timestamp (${source1.rtpTimestamp}`);
+    // Check that a received RTP audio level 127 is exactly 0
+    const source2 = contributingSources.find(c => c.source == csrc2);
+    ok(source2, "third csrc was found");
+    is(source2.audioLevel, 0,
+      `Contributing source has audio level of 0 when RTP audio level is 127`);
+    // Check caching
+    is(JSON.stringify(contributingSources),
+       JSON.stringify(remoteReceiver.getContributingSources()),
+       "getContributingSources is cached");
+    // Check that sources are sorted in descending order by time stamp
+    const timestamp3 =  performance.now() + performance.timeOrigin;
+    const rtpTimestamp3 = 44444;
+    // Larger offsets are further back in time
+    const testOffsets = [3, 7, 5, 6, 1, 4];
+    for (const offset of testOffsets) {
+      SpecialPowers.wrap(localReceiver).mozInsertAudioLevelForContributingSource(
+          offset, // Using offset for SSRC for convenience
+          timestamp3 - offset,
+          rtpTimestamp3,
+          true,
+          offset);
+    }
+    const sources = localReceiver.getContributingSources();
+    const sourceOffsets = sources.map(s => s.source);
+    is(JSON.stringify(sourceOffsets),
+       JSON.stringify([...testOffsets].sort((a, b) => a - b)),
+          `Contributing sources are sorted in descending order by timestamp:`
+          + ` ${JSON.stringify(sources)}`);
+  };
+
+  var test;
+  runNetworkTest(async function(options) {
+    test = new PeerConnectionTest(options);
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      [testGetContributingSources]);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    test.pcLocal.audioElementsOnly = true;
+    await pushPrefs(["privacy.reduceTimerPrecision", false]);
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioRenegotiationInactiveAnswer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioRenegotiationInactiveAnswer.html
new file mode 100644
index 0000000000..6d3a23b57a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioRenegotiationInactiveAnswer.html
@@ -0,0 +1,69 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="sdpUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1213773",
+    title: "Renegotiation: answerer uses a=inactive for audio"
+  });
+
+  runNetworkTest(function (options) {
+    const helper = new AudioStreamHelper();
+
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], []);
+    let haveFirstUnmuteEvent;
+
+    test.chain.insertBefore("PC_REMOTE_SET_LOCAL_DESCRIPTION", [
+      function PC_REMOTE_SETUP_ONUNMUTE_1() {
+        haveFirstUnmuteEvent = haveEvent(test.pcRemote._pc.getReceivers()[0].track, "unmute");
+      }
+    ]);
+
+    test.chain.append([
+      function PC_REMOTE_CHECK_AUDIO_UNMUTED() {
+        return haveFirstUnmuteEvent;
+      },
+      function PC_REMOTE_CHECK_AUDIO_FLOWING() {
+        return helper.checkAudioFlowing(test.pcRemote._pc.getRemoteStreams()[0]);
+      }
+    ]);
+
+    addRenegotiation(test.chain, []);
+
+    test.chain.insertAfter("PC_LOCAL_GET_ANSWER", [
+        function PC_LOCAL_REWRITE_REMOTE_SDP_INACTIVE(test) {
+          test._remote_answer.sdp =
+            sdputils.setAllMsectionsInactive(test._remote_answer.sdp);
+        }
+    ], false, 1);
+
+    test.chain.append([
+      function PC_REMOTE_CHECK_AUDIO_NOT_FLOWING() {
+        return helper.checkAudioNotFlowing(test.pcRemote._pc.getRemoteStreams()[0]);
+      }
+    ]);
+
+    test.chain.remove("PC_REMOTE_CHECK_STATS", 1);
+    test.chain.remove("PC_LOCAL_CHECK_STATS", 1);
+    test.chain.remove("PC_REMOTE_WAIT_FOR_MEDIA_FLOW", 1);
+
+    addRenegotiation(test.chain, []);
+
+    test.chain.append([
+      function PC_REMOTE_CHECK_AUDIO_FLOWING_2() {
+        return helper.checkAudioFlowing(test.pcRemote._pc.getRemoteStreams()[0]);
+      }
+    ]);
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioSynchronizationSources.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioSynchronizationSources.html
new file mode 100644
index 0000000000..32603b2e40
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioSynchronizationSources.html
@@ -0,0 +1,95 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1363667",
+    title: "Test audio receiver getSynchronizationSources"
+  });
+
+  var waitForSyncSources = async (test) => {
+    let receivers = [...test.pcRemote.getReceivers(),
+                     ...test.pcLocal.getReceivers()];
+    is(receivers.length, 2, "Expected number of receivers");
+    // Wait for sync sources
+    while (true) {
+        if (receivers[0].getSynchronizationSources().length &&
+            receivers[1].getSynchronizationSources().length) {
+        break;
+      }
+      await wait(250);
+    }
+  };
+
+  var testGetSynchronizationSources = async (test) => {
+    await waitForSyncSources(test);
+    let receivers = [...test.pcRemote.getReceivers(),
+                     ...test.pcLocal.getReceivers()];
+    is(receivers.length, 2,
+       `Expected number of receivers is 2. (${receivers.length})`);
+    for (let recv of receivers) {
+      let syncSources = recv.getSynchronizationSources();
+      ok(syncSources,
+         "Receiver has Synchronization sources " + JSON.stringify(syncSources));
+      is(syncSources.length, 1, "Each receiver has only a single sync source");
+      let source = recv.getSynchronizationSources()[0];
+      ok(source.audioLevel !== null,
+         `Synchronization source has audio level. (${source.audioLevel})`);
+      ok(source.audioLevel >= 0.0,
+         `Synchronization source audio level >= 0.0 (${source.audioLevel})`);
+      ok(source.audioLevel <= 1.0,
+         `Synchronization source audio level <= 1.0 (${source.audioLevel})`);
+      ok(source.timestamp,
+         `Synchronization source has timestamp (${source.timestamp})`);
+      const ageSeconds =
+        (window.performance.now() + window.performance.timeOrigin -
+         source.timestamp) / 1000;
+      ok(ageSeconds >= 0,
+         `Synchronization source timestamp is in the past`);
+      ok(ageSeconds < 2.5,
+         `Synchronization source timestamp is close to now`);
+      is(source.voiceActivityFlag, undefined,
+        "Synchronization source unsupported voiceActivity is undefined");
+    }
+  };
+
+  var testSynchronizationSourceCached = async (test) => {
+    await waitForSyncSources(test);
+    let receivers = [...test.pcRemote.getReceivers(),
+                     ...test.pcLocal.getReceivers()];
+    is(receivers.length, 2,
+       `Expected number of receivers is 2. (${receivers.length})`);
+    let sourceSets = [[],[]];
+    for (let sourceSet of sourceSets) {
+      for (let recv of receivers) {
+        let sources = recv.getSynchronizationSources();
+        is(sources.length, 1,
+           `Expected number of sources is 1. (${sources.length})`);
+        sourceSet.push(sources);
+      }
+      // Busy wait 1s before trying again
+      let endTime = performance.now() + 1000;
+      while (performance.now() < endTime) {};
+    }
+    is(JSON.stringify(sourceSets[0]), JSON.stringify(sourceSets[1]),
+       "Subsequent getSynchronizationSources calls are cached.");
+  };
+
+  var test;
+  runNetworkTest(function(options) {
+    test = new PeerConnectionTest(options);
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      [testGetSynchronizationSources,
+       testSynchronizationSourceCached]);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    test.pcLocal.audioElementsOnly = true;
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioSynchronizationSourcesUnidirectional.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioSynchronizationSourcesUnidirectional.html
new file mode 100644
index 0000000000..6d66614e91
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioSynchronizationSourcesUnidirectional.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1439001",
+    title: "Test audio unidirectional getSynchronizationSources"
+  });
+
+  var waitForSyncSources = async (test) => {
+    let receiver = test.pcRemote.getReceivers()[0];
+    ok(receiver, "Remote has a receiver");
+    // Wait for remote sync source
+    while (!receiver.getSynchronizationSources().length) {
+      await wait(250);
+    }
+    is(receiver.getSynchronizationSources().length, 1,
+       "Remote receiver has a synchronization source");
+    // Make sure local has no sync source
+    is(test.pcLocal.getReceivers()[0].getSynchronizationSources().length, 0,
+       "Local receiver has no synchronization source");
+  };
+  /*
+   * Test to make sure that in unidirectional calls, the receiving end has
+   * synchronization sources with audio levels, and the sending end has none.
+   */
+  var testGetSynchronizationSourcesUnidirectional = async (test) => {
+    await waitForSyncSources(test);
+    let receiver = test.pcRemote.getReceivers()[0];
+    let syncSources = receiver.getSynchronizationSources();
+    ok(syncSources.length,
+       "Receiver has Synchronization sources " + JSON.stringify(syncSources));
+    is(syncSources.length, 1, "Receiver has only a single sync source");
+    let syncSource = syncSources[0];
+    ok(syncSource.audioLevel !== undefined, "SynchronizationSource has audioLevel");
+  };
+
+  var test;
+  runNetworkTest(function(options) {
+    test = new PeerConnectionTest(options);
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      [testGetSynchronizationSourcesUnidirectional]);
+    test.setMediaConstraints([{audio: true}], []);
+    test.pcLocal.audioElementsOnly = true;
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio.html
new file mode 100644
index 0000000000..5fd10a67f9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796892",
+    title: "Basic audio-only peer connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    // pc.js uses video elements by default, we want to test audio elements here
+    test.pcLocal.audioElementsOnly = true;
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioDynamicPtMissingRtpmap.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioDynamicPtMissingRtpmap.html
new file mode 100644
index 0000000000..a076bf80f1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioDynamicPtMissingRtpmap.html
@@ -0,0 +1,36 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1246011",
+    title: "Offer with dynamic PT but missing rtpmap"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    // we want Opus to get selected and 101 to be ignored
+    options.opus = true;
+    test = new PeerConnectionTest(options);
+    test.chain.insertBefore("PC_REMOTE_GET_OFFER", [
+      function PC_LOCAL_REDUCE_MLINE_REMOVE_RTPMAPS(test) {
+        test.originalOffer.sdp =
+          sdputils.reduceAudioMLineToDynamicPtAndOpus(test.originalOffer.sdp);
+        test.originalOffer.sdp =
+          sdputils.removeAllRtpMaps(test.originalOffer.sdp);
+        test.originalOffer.sdp = test.originalOffer.sdp + "a=rtpmap:109 opus/48000/2\r\n";
+        info("SDP with dyn PT and no Rtpmap: " + JSON.stringify(test.originalOffer));
+      }
+    ]);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelay.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelay.html
new file mode 100644
index 0000000000..180abc075a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelay.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1231975",
+  title: "Basic audio-only peer connection with port dependent NAT, for verifying UDP relay"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_tcp', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "srflx";
+    options.expectedRemoteCandidateType = "relay";
+    // If both have TURN, it is a toss-up which one will end up using a
+    // relay.
+    options.turn_disabled_local = true;
+    const test = new PeerConnectionTest(options);
+    // Make sure we don't end up choosing the wrong thing due to delays in
+    // trickle. Once we are willing to accept trickle after ICE success, we
+    // can maybe wait a bit to allow things to stabilize.
+    // TODO(bug 1238249)
+    makeOffererNonTrickle(test.chain);
+    makeAnswererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTCP.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTCP.html
new file mode 100644
index 0000000000..7bb51764bd
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTCP.html
@@ -0,0 +1,42 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1231975",
+  title: "Basic audio-only peer connection with port dependent NAT that blocks UDP"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_udp', true],
+        ['media.peerconnection.nat_simulator.block_tcp', false],
+        ['media.peerconnection.nat_simulator.block_tls', true],
+        ['media.peerconnection.ice.loopback', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "relay-tcp";
+    options.expectedRemoteCandidateType = "relay-tcp";
+    // No reason to wait for gathering to complete like the other NAT tests,
+    // since relayed-tcp is the only thing that can work.
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTCPWithStun300.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTCPWithStun300.html
new file mode 100644
index 0000000000..43ea6aaea7
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTCPWithStun300.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "857668",
+  title: "Basic audio-only peer connection with UDP-blocking NAT, for verifying TCP relay with STUN 300 responses"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_udp', true],
+        ['media.peerconnection.nat_simulator.block_tcp', false],
+        ['media.peerconnection.ice.loopback', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "relay-tcp";
+    options.expectedRemoteCandidateType = "relay-tcp";
+    const turnServer = iceServersArray.find(server => "username" in server);
+    const turnRedirectPort = turnServer.turn_redirect_port;
+    const turnHostname = getTurnHostname(turnServer.urls[0]);
+    turnServer.urls = [`turn:${turnHostname}:${turnRedirectPort}?transport=tcp`];
+    // Override turn servers so we can test simulated redirects
+    options.config_local = {iceServers: [turnServer]};
+    options.config_remote = {iceServers: [turnServer]};
+    const test = new PeerConnectionTest(options);
+    // Make sure we don't end up choosing the wrong thing due to delays in
+    // trickle. Once we are willing to accept trickle after ICE success, we
+    // can maybe wait a bit to allow things to stabilize.
+    // TODO(bug 1238249)
+    makeOffererNonTrickle(test.chain);
+    makeAnswererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+    await SpecialPowers.popPrefEnv();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTLS.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTLS.html
new file mode 100644
index 0000000000..7446401f87
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTLS.html
@@ -0,0 +1,41 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1231975",
+  title: "Basic audio-only peer connection with port dependent NAT that blocks STUN"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_udp', true],
+        ['media.peerconnection.nat_simulator.block_tcp', true],
+        ['media.peerconnection.ice.loopback', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "relay-tls";
+    options.expectedRemoteCandidateType = "relay-tls";
+    // No reason to wait for gathering to complete like the other NAT tests,
+    // since relayed-tcp is the only thing that can work.
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayWithStun300.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayWithStun300.html
new file mode 100644
index 0000000000..286e67bc2f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayWithStun300.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "857668",
+  title: "Basic audio-only peer connection with port dependent NAT, for verifying UDP relay with STUN 300 responses"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_tcp', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "srflx";
+    options.expectedRemoteCandidateType = "relay";
+    const turnServer = iceServersArray.find(server => "username" in server);
+    const turnRedirectPort = turnServer.turn_redirect_port;
+    const turnHostname = getTurnHostname(turnServer.urls[0]);
+    turnServer.urls = [`turn:${turnHostname}:${turnRedirectPort}`];
+    // Override turn servers so we can test redirects
+    options.config_remote = {iceServers: [turnServer]};
+    // If both have TURN, it is a toss-up which one will end up using a
+    // relay, so we disable TURN for one side.
+    options.turn_disabled_local = true;
+    const test = new PeerConnectionTest(options);
+    // Make sure we don't end up choosing the wrong thing due to delays in
+    // trickle. Once we are willing to accept trickle after ICE success, we
+    // can maybe wait a bit to allow things to stabilize.
+    // TODO(bug 1238249)
+    makeOffererNonTrickle(test.chain);
+    makeAnswererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATSrflx.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATSrflx.html
new file mode 100644
index 0000000000..78fa8bcb2c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATSrflx.html
@@ -0,0 +1,44 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1231975",
+  title: "Basic audio-only peer connection with endpoint independent NAT, for verifying UDP srflx"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_tcp', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "srflx";
+    options.expectedRemoteCandidateType = "srflx";
+    const test = new PeerConnectionTest(options);
+    // Make sure we don't end up choosing the wrong thing due to delays in
+    // trickle. Once we are willing to accept trickle after ICE success, we
+    // can maybe wait a bit to allow things to stabilize.
+    // TODO(bug 1238249)
+    makeOffererNonTrickle(test.chain);
+    makeAnswererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNoisyUDPBlock.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNoisyUDPBlock.html
new file mode 100644
index 0000000000..297121cd94
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNoisyUDPBlock.html
@@ -0,0 +1,41 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1231975",
+  title: "Basic audio-only peer connection where UDP sockets return errors on send"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_udp', true],
+        ['media.peerconnection.nat_simulator.error_code_for_drop', 3 /*R_INTERNAL*/],
+        ['media.peerconnection.nat_simulator.block_tls', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "relay-tcp";
+    options.expectedRemoteCandidateType = "relay-tcp";
+    // No reason to wait for gathering to complete like the other NAT tests,
+    // since relayed-tcp is the only thing that can work.
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioPcmaPcmuOnly.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioPcmaPcmuOnly.html
new file mode 100644
index 0000000000..f0fe721b8e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioPcmaPcmuOnly.html
@@ -0,0 +1,39 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1221837",
+    title: "Only offer PCMA and PMCU in mline (no rtpmaps)"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.opus = false;
+    test = new PeerConnectionTest(options);
+    test.chain.insertBefore("PC_REMOTE_GET_OFFER", [
+      function PC_LOCAL_REDUCE_MLINE_REMOVE_RTPMAPS(test) {
+        test.originalOffer.sdp =
+          sdputils.reduceAudioMLineToPcmuPcma(test.originalOffer.sdp);
+        test.originalOffer.sdp =
+          sdputils.removeAllRtpMaps(test.originalOffer.sdp);
+        info("SDP without Rtpmaps: " + JSON.stringify(test.originalOffer));
+      }
+    ]);
+    test.chain.insertAfter("PC_REMOTE_SANE_LOCAL_SDP", [
+      function PC_REMOTE_VERIFY_PCMU(test) {
+        ok(test._remote_answer.sdp.includes("a=rtpmap:0 PCMU/8000"), "PCMU codec is present in SDP");
+      }
+    ]);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioRelayPolicy.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioRelayPolicy.html
new file mode 100644
index 0000000000..ced57ff8a3
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioRelayPolicy.html
@@ -0,0 +1,83 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1663746",
+  title: "Basic tests for relay ice policy"
+});
+
+runNetworkTest(async () => {
+  await pushPrefs(
+      // Enable mDNS, since there are some checks we want to run with that
+      ['media.peerconnection.ice.obfuscate_host_addresses', true]);
+
+  const offerer = new RTCPeerConnection({iceServers: iceServersArray, iceTransportPolicy: 'relay'});
+  const answerer = new RTCPeerConnection({iceServers: iceServersArray});
+
+  offerer.onicecandidate = e => {
+    if (e.candidate) {
+      ok(!e.candidate.candidate.includes(' host '), 'IceTransportPolicy \"relay\" should prevent the advertisement of host candidates');
+      ok(!e.candidate.candidate.includes(' srflx '), 'IceTransportPolicy \"relay\" should prevent the advertisement of srflx candidates');
+    }
+    answerer.addIceCandidate(e.candidate);
+  };
+
+  answerer.onicecandidate = e => {
+    if (e.candidate && e.candidate.candidate.includes(' host ')) {
+      ok(e.candidate.candidate.includes('.local'), 'When obfuscate_host_addresses is true, we expect host candidates to use mDNS');
+    }
+    offerer.addIceCandidate(e.candidate);
+  };
+
+  const offererConnected = new Promise(r => {
+    offerer.oniceconnectionstatechange = () => {
+      if (offerer.iceConnectionState == 'connected') {
+        r();
+      }
+    };
+  });
+
+  const answererConnected = new Promise(r => {
+    answerer.oniceconnectionstatechange = () => {
+      if (answerer.iceConnectionState == 'connected') {
+        r();
+      }
+    };
+  });
+
+  const offer = await offerer.createOffer({offerToReceiveAudio: true});
+  await Promise.all([offerer.setLocalDescription(offer), answerer.setRemoteDescription(offer)]);
+  const answer = await answerer.createAnswer();
+  await Promise.all([answerer.setLocalDescription(answer), offerer.setRemoteDescription(answer)]);
+
+  info('Waiting for ICE to connect');
+  await Promise.all([offererConnected, answererConnected]);
+
+  const offererStats = await offerer.getStats();
+  const localCandidates = [...offererStats.values()].filter(stat => stat.type == 'local-candidate');
+  const remoteCandidates = [...offererStats.values()].filter(stat => stat.type == 'remote-candidate');
+  isnot(localCandidates, []);
+  isnot(remoteCandidates, []);
+
+  const localNonRelayCandidates =
+      localCandidates.filter(cand => cand.candidateType != 'relay');
+  is(localNonRelayCandidates.length, 0, `There should only be local relay candidates, because we are using the "relay" IceTransportPolicy, but we got ${JSON.stringify(localNonRelayCandidates)}`);
+
+  const remoteHostCandidates =
+      remoteCandidates.filter(cand => cand.candidateType == 'host');
+  is(remoteHostCandidates.length, 0, `There should be no remote host candidates in the stats, because mDNS resolution should have been disabled by the "relay" IceTransportPolicy, but we got ${JSON.stringify(remoteHostCandidates)}`);
+
+  offerer.close();
+  answerer.close();
+
+  await SpecialPowers.popPrefEnv();
+}, { useIceServer: true });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioRequireEOC.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioRequireEOC.html
new file mode 100644
index 0000000000..afad4550d4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioRequireEOC.html
@@ -0,0 +1,35 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1167443",
+    title: "Basic audio-only peer connection which waits for end-of-candidates"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.chain.replace("PC_LOCAL_VERIFY_SDP_AFTER_END_OF_TRICKLE", [
+      function PC_LOCAL_REQUIRE_SDP_AFTER_END_OF_TRICKLE(test) {
+        return test.pcLocal.endOfTrickleSdp.then(sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(sdp, test.testOptions, test.pcLocal.label));
+      }
+    ]);
+    test.chain.replace("PC_REMOTE_VERIFY_SDP_AFTER_END_OF_TRICKLE", [
+      function PC_REMOTE_REQUIRE_SDP_AFTER_END_OF_TRICKLE(test) {
+        return test.pcRemote.endOfTrickleSdp.then(sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(sdp, test.testOptions, test.pcRemote.label));
+      }
+    ]);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVerifyRtpHeaderExtensions.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVerifyRtpHeaderExtensions.html
new file mode 100644
index 0000000000..f28a990bd2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVerifyRtpHeaderExtensions.html
@@ -0,0 +1,63 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="parser_rtp.js"></script>
+  <script type="application/javascript" src="sdpUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1416932",
+    title: "Basic audio-only peer connection and verify rtp header extensions"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    // pc.js uses video elements by default, we want to test audio elements here
+    test.pcLocal.audioElementsOnly = true;
+
+    let getRtpPacket = (pc) => {
+      // we only examine received packets
+      let sending = false;
+      pc.mozEnablePacketDump(0, "rtp", sending);
+      return new Promise((res, rej) =>
+        pc.mozSetPacketCallback((...args) => {
+          res([...args]);
+          pc.mozSetPacketCallback(() => {});
+          pc.mozDisablePacketDump(0, "rtp", sending);
+        })
+      );
+    }
+
+    const pc = SpecialPowers.wrap(test.pcRemote._pc);
+    const haveFirstPacket = getRtpPacket(pc);
+
+    test.chain.insertBefore('PC_REMOTE_WAIT_FOR_MEDIA_FLOW', [
+      async function PC_REMOTE_CHECK_RTP_HEADER_EXTS_AGAINST_SDP() {
+
+        const sdpExtmapIds = sdputils.findExtmapIds(test.originalAnswer.sdp);
+
+        const [level, type, sending, data] = await haveFirstPacket;
+        const extensions = ParseRtpPacket(data).header.extensions;
+
+        // make sure we got the same number of rtp header extensions in
+        // the received packet as were negotiated in the sdp.  Then
+        // check to make sure each of the received extension ids were in
+        // the sdp.
+        is(sdpExtmapIds.length, extensions.length, "number of received ids match sdp ids");
+        // note, we are comparing a number (from the parsed rtp packet)
+        // and a string (from the answer sdp)
+        ok(extensions.every((ext) => sdpExtmapIds.includes(""+ext.id)), "extension id arrays equivalent");
+      }
+    ]);
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideo.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideo.html
new file mode 100644
index 0000000000..c2c2d43f09
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideo.html
@@ -0,0 +1,24 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796890",
+    title: "Basic audio/video (separate) peer connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoCombined.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoCombined.html
new file mode 100644
index 0000000000..02a561f9b8
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoCombined.html
@@ -0,0 +1,24 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796890",
+    title: "Basic audio/video (combined) peer connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true, video: true}],
+                             [{audio: true, video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoBundle.html
new file mode 100644
index 0000000000..cae7f6617f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoBundle.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1016476",
+    title: "Basic audio/video peer connection with no Bundle"
+  });
+
+  runNetworkTest(options => {
+    options = options || { };
+    options.bundle = false;
+    var test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoBundleNoRtcpMux.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoBundleNoRtcpMux.html
new file mode 100644
index 0000000000..49b0136752
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoBundleNoRtcpMux.html
@@ -0,0 +1,39 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1167443",
+    title: "Basic audio & video call with disabled bundle and disabled RTCP-Mux"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.bundle = false;
+    options.rtcpmux = false;
+    test = new PeerConnectionTest(options);
+    test.chain.replace("PC_LOCAL_VERIFY_SDP_AFTER_END_OF_TRICKLE", [
+      function PC_LOCAL_REQUIRE_SDP_AFTER_END_OF_TRICKLE(test) {
+        return test.pcLocal.endOfTrickleSdp .then(sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(sdp, test.testOptions, test.pcLocal.label));
+      }
+    ]);
+    test.chain.replace("PC_REMOTE_VERIFY_SDP_AFTER_END_OF_TRICKLE", [
+      function PC_REMOTE_REQUIRE_SDP_AFTER_END_OF_TRICKLE(test) {
+        return test.pcRemote.endOfTrickleSdp .then(sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(sdp, test.testOptions, test.pcRemote.label));
+      }
+    ]);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoRtcpMux.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoRtcpMux.html
new file mode 100644
index 0000000000..48524604ba
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoRtcpMux.html
@@ -0,0 +1,38 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1167443",
+    title: "Basic audio & video call with disabled RTCP-Mux"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.rtcpmux = false;
+    test = new PeerConnectionTest(options);
+    test.chain.replace("PC_LOCAL_VERIFY_SDP_AFTER_END_OF_TRICKLE", [
+      function PC_LOCAL_REQUIRE_SDP_AFTER_END_OF_TRICKLE(test) {
+        return test.pcLocal.endOfTrickleSdp .then(sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(sdp, test.testOptions, test.pcLocal.label));
+      }
+    ]);
+    test.chain.replace("PC_REMOTE_VERIFY_SDP_AFTER_END_OF_TRICKLE", [
+      function PC_REMOTE_REQUIRE_SDP_AFTER_END_OF_TRICKLE(test) {
+        return test.pcRemote.endOfTrickleSdp .then(sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(sdp, test.testOptions, test.pcRemote.label));
+      }
+    ]);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoTransceivers.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoTransceivers.html
new file mode 100644
index 0000000000..181d089d26
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoTransceivers.html
@@ -0,0 +1,31 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1290948",
+    title: "Basic audio/video with addTransceiver"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    test.chain.replace("PC_LOCAL_GUM",
+      [
+        function PC_LOCAL_GUM_TRANSCEIVERS(test) {
+          return test.pcLocal.getAllUserMediaAndAddTransceivers(test.pcLocal.constraints);
+        }
+      ]);
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyExtmap.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyExtmap.html
new file mode 100644
index 0000000000..e3da00bfa5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyExtmap.html
@@ -0,0 +1,97 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1406529",
+    title: "Verify SDP extmap attribute for sendrecv connection"
+  });
+
+  var test;
+  runNetworkTest(async function (options) {
+    await pushPrefs(["media.navigator.video.use_transport_cc", true]);
+
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+
+    test.chain.insertAfter('PC_LOCAL_SET_LOCAL_DESCRIPTION', [
+      async function PC_LOCAL_CHECK_SDP_OFFER_EXTMAP() {
+        sdputils.verify_unique_extmap_ids(test.originalOffer.sdp);
+
+        const audio = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getAudioMSections(test.originalOffer.sdp));
+        const expected_audio = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["1", "urn:ietf:params:rtp-hdrext:ssrc-audio-level", ""],
+          ["2", "urn:ietf:params:rtp-hdrext:csrc-audio-level", "recvonly"],
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid", ""],
+        ];
+        // *Ugh* ...
+        ok(JSON.stringify(audio) ===
+           JSON.stringify(expected_audio),
+           "List of offer audio URNs meets expected values");
+
+        const video = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getVideoMSections(test.originalOffer.sdp));
+        const expected_video = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid", ""],
+          ["4", "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", ""],
+          ["5", "urn:ietf:params:rtp-hdrext:toffset", ""],
+          ["6", "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", "recvonly"],
+          ["7", "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", ""],
+        ];
+        // *Ugh* ...
+        ok(JSON.stringify(video) ===
+           JSON.stringify(expected_video),
+           "List of offer video URNs meets expected values");
+      }
+    ]);
+
+    test.chain.removeAfter('PC_REMOTE_SET_LOCAL_DESCRIPTION');
+    test.chain.append([
+      async function PC_REMOTE_CHECK_SDP_ANSWER_EXTMAP() {
+        sdputils.verify_unique_extmap_ids(test.originalAnswer.sdp);
+
+        const audio = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getAudioMSections(test.originalAnswer.sdp));
+        const expected_audio = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["1", "urn:ietf:params:rtp-hdrext:ssrc-audio-level",""],
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid",""],
+        ];
+        // *Ugh* ...
+        ok(JSON.stringify(audio) ===
+           JSON.stringify(expected_audio),
+           "List of answer audio URNs meets expected values");
+
+        const video = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getVideoMSections(test.originalAnswer.sdp));
+        const expected_video = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid",""],
+          ["4", "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",""],
+          ["5", "urn:ietf:params:rtp-hdrext:toffset",""],
+          ["7", "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", ""],
+        ];
+        ok(JSON.stringify(video) ===
+           JSON.stringify(expected_video),
+           "List of answer video URNs meets expected values");
+      }
+    ]);
+
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyExtmapSendonly.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyExtmapSendonly.html
new file mode 100644
index 0000000000..6cbc9e4c00
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyExtmapSendonly.html
@@ -0,0 +1,97 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1406529",
+    title: "Verify SDP extmap attribute for sendonly connection"
+  });
+
+  var test;
+  runNetworkTest(async function (options) {
+    await pushPrefs(["media.navigator.video.use_transport_cc", true]);
+
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             []);
+
+    test.chain.insertAfter('PC_LOCAL_SET_LOCAL_DESCRIPTION', [
+      async function PC_LOCAL_CHECK_SDP_OFFER_EXTMAP() {
+        sdputils.verify_unique_extmap_ids(test.originalOffer.sdp);
+
+        const audio = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getAudioMSections(test.originalOffer.sdp));
+        const expected_audio = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["1", "urn:ietf:params:rtp-hdrext:ssrc-audio-level", ""],
+          ["2", "urn:ietf:params:rtp-hdrext:csrc-audio-level", "recvonly"],
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid", ""],
+        ];
+        // *Ugh* ...
+        ok(JSON.stringify(audio) ===
+           JSON.stringify(expected_audio),
+           "List of offer audio URNs meets expected values");
+
+        const video = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getVideoMSections(test.originalOffer.sdp));
+        const expected_video = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid", ""],
+          ["4", "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", ""],
+          ["5", "urn:ietf:params:rtp-hdrext:toffset", ""],
+          ["6", "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", "recvonly"],
+          ["7", "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", ""],
+        ];
+        // *Ugh* ...
+        ok(JSON.stringify(video) ===
+           JSON.stringify(expected_video),
+           "List of offer video URNs meets expected values");
+      }
+    ]);
+
+    test.chain.removeAfter('PC_REMOTE_SET_LOCAL_DESCRIPTION');
+    test.chain.append([
+      async function PC_REMOTE_CHECK_SDP_ANSWER_EXTMAP() {
+        sdputils.verify_unique_extmap_ids(test.originalAnswer.sdp);
+
+        const audio = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getAudioMSections(test.originalAnswer.sdp));
+        const expected_audio = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["1", "urn:ietf:params:rtp-hdrext:ssrc-audio-level",""],
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid",""],
+        ];
+        // *Ugh* ...
+        ok(JSON.stringify(audio) ===
+           JSON.stringify(expected_audio),
+           "List of answer audio URNs meets expected values");
+
+        const video = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getVideoMSections(test.originalAnswer.sdp));
+        const expected_video = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid",""],
+          ["4", "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",""],
+          ["5", "urn:ietf:params:rtp-hdrext:toffset",""],
+          ["7", "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", ""],
+        ];
+        ok(JSON.stringify(video) ===
+           JSON.stringify(expected_video),
+           "List of answer video URNs meets expected values");
+      }
+    ]);
+
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyTooLongMidFails.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyTooLongMidFails.html
new file mode 100644
index 0000000000..70d27b48c6
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyTooLongMidFails.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1427009",
+    title: "Test mid longer than 16 characters fails"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.bundle = false;
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+
+    test.chain.replaceAfter("PC_LOCAL_CREATE_OFFER",
+      [
+        function PC_LOCAL_MUNGE_OFFER_SDP(test) {
+          test.originalOffer.sdp =
+            test.originalOffer.sdp.replace(/a=mid:.*\r\n/g,
+                                           "a=mid:really_long_mid_over_16_chars\r\n");
+        },
+        function PC_LOCAL_EXPECT_SET_LOCAL_DESCRIPTION_FAIL(test) {
+          return test.setLocalDescription(test.pcLocal,
+                                          test.originalOffer,
+                                          HAVE_LOCAL_OFFER)
+           .then(() => ok(false, "setLocalDescription must fail"),
+                 // This needs to be RTCError once we support it, and once we
+                 // stop allowing any modification, InvalidModificationError
+                 e => is(e.name, "OperationError",
+                         "setLocalDescription must fail and did"));
+        }
+      ], 0 // first occurance
+    );
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio_forced_higher_rate.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio_forced_higher_rate.html
new file mode 100644
index 0000000000..95bfb06514
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio_forced_higher_rate.html
@@ -0,0 +1,19 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="peerconnection_audio_forced_sample_rate.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1437366",
+  title: "Basic audio-only peer connection, with the MTG running at a rate not supported by the MediaPipeline (49000Hz)"
+});
+
+test_peerconnection_audio_forced_sample_rate(49000);
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio_forced_lower_rate.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio_forced_lower_rate.html
new file mode 100644
index 0000000000..aab9778971
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio_forced_lower_rate.html
@@ -0,0 +1,19 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="peerconnection_audio_forced_sample_rate.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1437366",
+  title: "Basic audio-only peer connection, with the MTG running at a rate not supported by the MediaPipeline (24000Hz)"
+});
+
+test_peerconnection_audio_forced_sample_rate(24000);
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicH264Video.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicH264Video.html
new file mode 100644
index 0000000000..072c35da39
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicH264Video.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1040346",
+    title: "Basic H.264 GMP video-only peer connection"
+  });
+
+  var test;
+  runNetworkTest(async function (options) {
+    matchPlatformH264CodecPrefs();
+    options = options || { };
+    options.h264 = true;
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicScreenshare.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicScreenshare.html
new file mode 100644
index 0000000000..93148ac5fd
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicScreenshare.html
@@ -0,0 +1,56 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1039666",
+    title: "Basic screenshare-only peer connection"
+  });
+
+  async function supportedVideoPayloadTypes() {
+    const pc = new RTCPeerConnection();
+    const offer = await pc.createOffer({offerToReceiveVideo: true});
+    return sdputils.getPayloadTypes(offer.sdp);
+  }
+
+  async function testScreenshare(payloadType) {
+    const options = {};
+    options.h264 = payloadType == 97 || payloadType == 126;
+    const test = new PeerConnectionTest(options);
+    const constraints = {
+      video: { mediaSource: "screen" },
+    };
+    test.setMediaConstraints([constraints], []);
+    test.chain.insertAfterEach("PC_LOCAL_CREATE_OFFER", [
+      function PC_LOCAL_ISOLATE_CODEC() {
+        info(`Forcing payload type ${payloadType}. Note that other associated ` +
+             `payload types, like RTX, are removed too.`);
+        test.originalOffer.sdp =
+          sdputils.removeAllButPayloadType(test.originalOffer.sdp, payloadType);
+      },
+    ]);
+    await test.run();
+  }
+
+  runNetworkTest(async () => {
+    await matchPlatformH264CodecPrefs();
+    const pts = await supportedVideoPayloadTypes();
+    ok(pts.includes("120"), "VP8 is supported");
+    ok(pts.includes("121"), "VP9 is supported");
+    if (pts.length > 2) {
+      is(pts.length, 4, "Expected VP8, VP9 and two variants of H264");
+      ok(pts.includes("97"), "H264 with no packetization-mode is supported");
+      ok(pts.includes("126"), "H264 with packetization-mode=1 is supported");
+    }
+    for (const pt of pts) {
+      await testScreenshare(pt);
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicVideo.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicVideo.html
new file mode 100644
index 0000000000..4a0655d696
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicVideo.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796888",
+    title: "Basic video-only peer connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicVideoVerifyRtpHeaderExtensions.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicVideoVerifyRtpHeaderExtensions.html
new file mode 100644
index 0000000000..7874e52a10
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicVideoVerifyRtpHeaderExtensions.html
@@ -0,0 +1,82 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="parser_rtp.js"></script>
+  <script type="application/javascript" src="sdpUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1416932",
+    title: "Basic video-only peer connection and verify rtp header extensions"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+
+    let getRtpPacketWithExtension = (pc, extension) => {
+      // we only examine received packets
+      let sending = false;
+      pc.mozEnablePacketDump(0, "rtp", sending);
+      return new Promise((res, rej) =>
+        pc.mozSetPacketCallback((...args) => {
+          const packet = ParseRtpPacket(args[3]);
+          info(`midId = ${extension} packet = ${JSON.stringify(packet, null, 2)}`);
+          if (packet.header.extensions.find(e => e.id == extension) !== undefined) {
+            res(packet);
+            pc.mozSetPacketCallback(() => {});
+            pc.mozDisablePacketDump(0, "rtp", sending);
+          }
+        })
+      );
+    }
+
+    let havePacketWithMid;
+    let sdpExtmaps;
+
+    // MID can stop being sent when acked causing failures if packets are checked later.
+    // Starting packet sniffer before PC_LOCAL_SET_REMOTE_DESCRIPTION to be ready
+    // to inspect packets ahead of any packets arriving.
+    test.chain.insertBefore('PC_LOCAL_SET_REMOTE_DESCRIPTION', [
+      function PC_REMOTE_FIND_RTP_PACKETS_WITH_MIDID() {
+
+        sdpExtmaps = sdputils.findExtmapIdsUrnsDirections(test.originalAnswer.sdp);
+        const [midId] = sdpExtmaps.find(([, urn]) => urn == "urn:ietf:params:rtp-hdrext:sdes:mid");
+        const pc = SpecialPowers.wrap(test.pcRemote._pc);
+        havePacketWithMid = getRtpPacketWithExtension(pc, midId);
+      }
+    ]);
+
+    test.chain.insertBefore('PC_REMOTE_WAIT_FOR_MEDIA_FLOW', [
+      async function PC_REMOTE_CHECK_RTP_HEADER_EXTS_AGAINST_SDP() {
+
+        const sdpExtmapIds = sdpExtmaps.map(e => e[0]);
+        const packet = await havePacketWithMid;
+        const extIds = packet.header.extensions.map(e => `${e.id}`);
+        // make sure we got the same number of rtp header extensions in
+        // the received packet as were negotiated in the sdp.  Then
+        // check to make sure each of the received extension ids were in
+        // the sdp.
+        is(sdpExtmapIds.length, extIds.length,
+           `number of sdp ids match received ids ` +
+           `${JSON.stringify(sdpExtmapIds)} == ${JSON.stringify(extIds)}\n` +
+           `sdp = ${test.originalAnswer.sdp}\n` +
+           `packet = ${JSON.stringify(packet, null, 2)}`);
+        // note, we are comparing a number (from the parsed rtp packet)
+        // and a string (from the answer sdp)
+        ok(extIds.every(id => sdpExtmapIds.includes(id)) &&
+           sdpExtmapIds.every(id => extIds.includes(id)),
+          `extension id arrays equivalent`);
+      }
+    ]);
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicWindowshare.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicWindowshare.html
new file mode 100644
index 0000000000..1cfb0797db
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicWindowshare.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1038926",
+    title: "Basic windowshare-only peer connection"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    const constraints = {
+      video: { mediaSource: "window" },
+    };
+    test.setMediaConstraints([constraints], []);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1013809.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1013809.html
new file mode 100644
index 0000000000..a8c7004793
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1013809.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1013809",
+    title: "Audio-only peer connection with swapped setLocal and setRemote steps"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    var sld = test.chain.remove("PC_REMOTE_SET_LOCAL_DESCRIPTION");
+    test.chain.insertAfter("PC_LOCAL_SET_REMOTE_DESCRIPTION", sld);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1042791.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1042791.html
new file mode 100644
index 0000000000..a84dcf9d09
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1042791.html
@@ -0,0 +1,36 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1040346",
+    title: "Basic H.264 GMP video-only peer connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.h264 = true;
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    test.chain.removeAfter("PC_LOCAL_CREATE_OFFER");
+
+    test.chain.append([
+      function PC_LOCAL_VERIFY_H264_OFFER(test) {
+        ok(!test.pcLocal._latest_offer.sdp.toLowerCase().includes("profile-level-id=0x42e0"),
+           "H264 offer does not contain profile-level-id=0x42e0");
+        ok(test.pcLocal._latest_offer.sdp.toLowerCase().includes("profile-level-id=42e0"),
+           "H264 offer contains profile-level-id=42e0");
+      }
+    ]);
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1227781.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1227781.html
new file mode 100644
index 0000000000..41e4aec457
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1227781.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1227781",
+    title: "Test with invalid TURN server"
+  });
+
+  const turnConfig = {
+    iceServers: [
+      {
+        username: "mozilla",
+        credential: "mozilla",
+        url: "turn:test@10.0.0.1",
+      },
+    ],
+  };
+  runNetworkTest(function (options) {
+    let exception = false;
+    try {
+      new RTCPeerConnection(turnConfig);
+    } catch (e) {
+      info(e);
+      exception = true;
+    }
+    is(exception, true, "Exception fired");
+    ok("Success");
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1512281.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1512281.html
new file mode 100644
index 0000000000..e6451becea
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1512281.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1512281",
+    title: "Test that RTCP sender and receiver stats are not swapped"
+  });
+
+const ensure_missing_rtcp = async stats => {
+	const rtcp_stats = [...stats.values()].filter(
+		s => s.type.endsWith("bound-rtp") &&
+			 s.isRemote == true).map(s => JSON.stringify(s))
+	is(rtcp_stats, [],
+	   "There are no RTCP stats when RTCP reception is turned off");
+};
+
+const PC_LOCAL_TEST_FOR_MISSING_RTCP = async test =>
+	await ensure_missing_rtcp(await test.pcLocal.getStats());
+
+const PC_REMOTE_TEST_FOR_MISSING_RTCP = async test =>
+  await ensure_missing_rtcp(await test.pcRemote.getStats());
+
+runNetworkTest(async options => {
+  await pushPrefs(["media.webrtc.net.force_disable_rtcp_reception", true]);
+
+  const test = new PeerConnectionTest(options);
+
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+    [PC_LOCAL_TEST_FOR_MISSING_RTCP]);
+
+  test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+    [PC_REMOTE_TEST_FOR_MISSING_RTCP]);
+
+  test.setMediaConstraints([{audio: true}, {video: true}],
+                           [{audio: true}, {video: true}]);
+  await test.run();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1773067.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1773067.html
new file mode 100644
index 0000000000..9e6d79a107
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1773067.html
@@ -0,0 +1,32 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1773067",
+    title: "getStats on a closed peer connection should fail, not hang, " +
+           " until bug 1056433 is fixed"
+  });
+
+  // TODO: Bug 1056433 removes the need for this test
+  runNetworkTest(async function () {
+    let errorName;
+    try {
+      const pc = new RTCPeerConnection();
+      pc.close();
+      await pc.getStats();
+    } catch(e) {
+      errorName = e.name;
+    }
+    is(errorName,
+       "InvalidStateError",
+       "getStats on closed peer connection fails instead of hanging");
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug822674.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug822674.html
new file mode 100644
index 0000000000..fceb2c2a1d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug822674.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "822674",
+    title: "RTCPeerConnection isn't a true javascript object as it should be"
+  });
+
+  runNetworkTest(function () {
+    var pc = new RTCPeerConnection();
+
+    pc.thereIsNeverGoingToBeAPropertyWithThisNameOnThisInterface = 1;
+    is(pc.thereIsNeverGoingToBeAPropertyWithThisNameOnThisInterface, 1,
+       "Can set expandos on an RTCPeerConnection");
+
+    pc = null;
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug825703.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug825703.html
new file mode 100644
index 0000000000..5cd168af8a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug825703.html
@@ -0,0 +1,140 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "825703",
+    title: "RTCConfiguration valid/invalid permutations"
+  });
+
+// ^^^ Don't insert data above this line without adjusting line number below!
+var lineNumberAndFunction = {
+// <--- 16 is the line this must be.
+  line: 17, func: () => new RTCPeerConnection().onaddstream = () => {}
+};
+
+var makePC = (config, expected_error) => {
+  var exception;
+  try {
+    new RTCPeerConnection(config).close();
+  } catch (e) {
+    exception = e;
+  }
+  is((exception? exception.name : "success"), expected_error || "success",
+     "RTCPeerConnection(" + JSON.stringify(config) + ")");
+};
+
+// The order of properties in objects is not guaranteed in JavaScript, so this
+// transform produces json-comparable dictionaries. The resulting copy is only
+// meant to be used in comparisons (e.g. array-ness is not preserved).
+
+var toComparable = o =>
+    (typeof o != 'object' || !o)? o : Object.keys(o).sort().reduce((co, key) => {
+  co[key] = toComparable(o[key]);
+  return co;
+}, {});
+
+// This is a test of the iceServers parsing code + readable errors
+runNetworkTest(() => {
+  var exception = null;
+
+  try {
+    new RTCPeerConnection().close();
+  } catch (e) {
+    exception = e;
+  }
+  ok(!exception, "RTCPeerConnection() succeeds");
+  exception = null;
+
+  // Some overlap still with WPT RTCConfiguration-iceServers.html
+
+  makePC({ iceServers: [
+    { urls:"stun:127.0.0.1" },
+    { urls:"stun:localhost", foo:"" },
+    { urls: ["stun:127.0.0.1", "stun:localhost"] },
+    { urls:"stuns:localhost", foo:"" },
+    { urls:"turn:[::1]:3478", username:"p", credential:"p" },
+    { urls:"turn:[::1]:3478", username:"", credential:"" },
+    { urls:"turns:[::1]:3478", username:"", credential:"" },
+    { urls:"turn:localhost:3478?transport=udp", username:"p", credential:"p" },
+    { urls: ["turn:[::1]:3478", "turn:localhost"], username:"p", credential:"p" },
+    { urls:"turns:localhost:3478?transport=udp", username:"p", credential:"p" },
+    { url:"stun:localhost", foo:"" },
+    { url:"turn:localhost", username:"p", credential:"p" }
+  ]});
+
+  makePC({ iceServers: [{ urls:"http:0.0.0.0" }] }, "SyntaxError");
+
+  try {
+    new RTCPeerConnection({ iceServers: [{ url:"http:0.0.0.0" }] }).close();
+  } catch (e) {
+    ok(e.message.indexOf("http") > 0,
+       "RTCPeerConnection() constructor has readable exceptions");
+  }
+
+  // Test getConfiguration
+  const config = {
+    bundlePolicy: "max-bundle",
+    iceTransportPolicy: "relay",
+    peerIdentity: null,
+    certificates: [],
+    iceServers: [
+      { urls: ["stun:127.0.0.1", "stun:localhost"], credentialType:"password" },
+      { urls: ["turn:[::1]:3478"], username:"p", credential:"p", credentialType:"password" },
+    ],
+  };
+  // Make sure sdpSemantics is not exposed in getConfiguration
+  const configWithExtraProps = Object.assign({},
+                                             config,
+                                             {sdpSemantics: "plan-b"});
+  ok("sdpSemantics" in configWithExtraProps, "sdpSemantics control");
+
+  const pc = new RTCPeerConnection(configWithExtraProps);
+  is(JSON.stringify(toComparable(pc.getConfiguration())),
+     JSON.stringify(toComparable(config)), "getConfiguration");
+  pc.close();
+
+  var push = prefs => SpecialPowers.pushPrefEnv(prefs);
+
+  return Promise.resolve()
+  // This set of tests are setting the about:config User preferences for default
+  // ice servers and checking the outputs when RTCPeerConnection() is
+  // invoked. See Bug 1167922 for more information.
+  .then(() => push({ set: [['media.peerconnection.default_iceservers', ""]] })
+    .then(() => makePC())
+    .then(() => push({ set: [['media.peerconnection.default_iceservers', "k"]] }))
+    .then(() => makePC())
+    .then(() => push({ set: [['media.peerconnection.default_iceservers', "[{\"urls\": [\"stun:stun.services.mozilla.com\"]}]"]] }))
+    .then(() => makePC()))
+  // This set of tests check that warnings work. See Bug 1254839 for more.
+  .then(() => {
+    let promise = new Promise(resolve => {
+      SpecialPowers.registerConsoleListener(msg => {
+        if (msg.message.includes("onaddstream")) {
+          SpecialPowers.postConsoleSentinel();
+          resolve(msg.message);
+        }
+      });
+    });
+    lineNumberAndFunction.func();
+    return promise;
+  }).then(warning => {
+    is(warning.split('"')[1],
+       "WebRTC: onaddstream is deprecated! Use peerConnection.ontrack instead.",
+       "warning logged");
+    var remainder = warning.split('"').slice(2).join('"');
+    info(remainder);
+    ok(remainder.includes('file: "' + window.location + '"'),
+       "warning has this file");
+    ok(remainder.includes('line: ' + lineNumberAndFunction.line),
+       "warning has correct line number");
+    });
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug827843.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug827843.html
new file mode 100644
index 0000000000..06cfde9e5d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug827843.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "827843",
+    title: "Ensure that localDescription and remoteDescription are null after close"
+  });
+
+var steps = [
+  function CHECK_SDP_ON_CLOSED_PC(test) {
+    var description;
+    var exception = null;
+
+    test.pcLocal.close();
+
+    try { description = test.pcLocal.localDescription; } catch (e) { exception = e; }
+    ok(exception, "Attempt to access localDescription of pcLocal after close throws exception");
+    exception = null;
+
+    try { description = test.pcLocal.remoteDescription; } catch (e) { exception = e; }
+    ok(exception, "Attempt to access remoteDescription of pcLocal after close throws exception");
+    exception = null;
+
+    test.pcRemote.close();
+
+    try  { description = test.pcRemote.localDescription; } catch (e) { exception = e; }
+    ok(exception, "Attempt to access localDescription of pcRemote after close throws exception");
+    exception = null;
+
+    try  { description = test.pcRemote.remoteDescription; } catch (e) { exception = e; }
+    ok(exception, "Attempt to access remoteDescription of pcRemote after close throws exception");
+  }
+];
+
+var test;
+runNetworkTest(() => {
+  test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.append(steps);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug834153.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug834153.html
new file mode 100644
index 0000000000..6d8ca2a7ce
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug834153.html
@@ -0,0 +1,36 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "834153",
+    title: "Queue CreateAnswer in PeerConnection.js"
+  });
+
+  runNetworkTest(function () {
+    var pc1 = new RTCPeerConnection();
+    var pc2 = new RTCPeerConnection();
+
+    return pc1.createOffer({ offerToReceiveAudio: true }).then(offer => {
+      // The whole point of this test is not to wait for the
+      // setRemoteDescription call to succesfully complete, so we
+      // don't wait for it to succeed.
+      pc2.setRemoteDescription(offer);
+      return pc2.createAnswer();
+    })
+    .then(answer => is(answer.type, "answer", "CreateAnswer created an answer"))
+    .catch(reason => ok(false, reason.message))
+    .then(() => {
+      pc1.close();
+      pc2.close();
+    })
+    .catch(reason => ok(false, reason.message));
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_callbacks.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_callbacks.html
new file mode 100644
index 0000000000..4c890e4400
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_callbacks.html
@@ -0,0 +1,86 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "PeerConnection using callback functions",
+    bug: "1119593",
+    visible: true
+  });
+
+// This still aggressively uses promises, but it is testing that the callback functions
+// are properly in place.
+
+// wrapper that turns a callback-based function call into a promise
+function pcall(o, f, beforeArg) {
+  return new Promise((resolve, reject) => {
+    var args = [resolve, reject];
+    if (typeof beforeArg !== 'undefined') {
+      args.unshift(beforeArg);
+    }
+    info('Calling ' + f.name);
+    f.apply(o, args);
+  });
+}
+
+var pc1 = new RTCPeerConnection();
+var pc2 = new RTCPeerConnection();
+
+var pc2_haveRemoteOffer = new Promise(resolve => {
+  pc2.onsignalingstatechange =
+    e => (e.target.signalingState == "have-remote-offer") && resolve();
+});
+var pc1_stable = new Promise(resolve => {
+  pc1.onsignalingstatechange =
+    e => (e.target.signalingState == "stable") && resolve();
+});
+
+pc1.onicecandidate = e => {
+  pc2_haveRemoteOffer
+    .then(() => !e.candidate || pcall(pc2, pc2.addIceCandidate, e.candidate))
+    .catch(generateErrorCallback());
+};
+pc2.onicecandidate = e => {
+  pc1_stable
+    .then(() => !e.candidate || pcall(pc1, pc1.addIceCandidate, e.candidate))
+    .catch(generateErrorCallback());
+};
+
+var v1, v2;
+var delivered = new Promise(resolve => {
+  pc2.onaddstream = e => {
+    v2.srcObject = e.stream;
+    resolve(e.stream);
+  };
+});
+
+runNetworkTest(function() {
+  v1 = createMediaElement('video', 'v1');
+  v2 = createMediaElement('video', 'v2');
+  var canPlayThrough = new Promise(resolve => v2.canplaythrough = resolve);
+  is(v2.currentTime, 0, "v2.currentTime is zero at outset");
+
+  // not testing legacy gUM here
+  return navigator.mediaDevices.getUserMedia({ video: true, audio: true })
+    .then(stream => pc1.addStream(v1.srcObject = stream))
+    .then(() => pcall(pc1, pc1.createOffer))
+    .then(offer => pcall(pc1, pc1.setLocalDescription, offer))
+    .then(() => pcall(pc2, pc2.setRemoteDescription, pc1.localDescription))
+    .then(() => pcall(pc2, pc2.createAnswer))
+    .then(answer => pcall(pc2, pc2.setLocalDescription, answer))
+    .then(() => pcall(pc1, pc1.setRemoteDescription, pc2.localDescription))
+    .then(() => delivered)
+  //    .then(() => canPlayThrough)    // why doesn't this fire?
+    .then(() => waitUntil(() => v2.currentTime > 0))
+    .then(() => ok(v2.currentTime > 0, "v2.currentTime is moving (" + v2.currentTime + ")"))
+    .then(() => ok(true, "Connected."))
+    .then(() => { v1.pause(); v2.pause(); });
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_2d.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_2d.html
new file mode 100644
index 0000000000..db3a735008
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_2d.html
@@ -0,0 +1,81 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1032848",
+  title: "Canvas(2D)::CaptureStream as video-only input to peerconnection",
+  visible: true
+});
+
+runNetworkTest(async () => {
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  var test = new PeerConnectionTest();
+  var mediaElement;
+  var h = new CaptureStreamTestHelper2D();
+  var canvas = document.createElement('canvas');
+  var stream;
+  canvas.id = 'source_canvas';
+  canvas.width = canvas.height = 16;
+  document.getElementById('content').appendChild(canvas);
+
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function PC_LOCAL_CANVAS_CAPTURESTREAM(test) {
+      h.drawColor(canvas, h.green);
+      stream = canvas.captureStream(0);
+      test.pcLocal.attachLocalStream(stream);
+      stream.requestFrame();
+      var i = 0;
+      return setInterval(function() {
+        try {
+          info("draw " + i ? "green" : "red");
+          h.drawColor(canvas, i ? h.green : h.red);
+          i = 1 - i;
+          stream.requestFrame();
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
+    }
+  ]);
+  test.chain.append([
+    function PC_REMOTE_WAIT_FOR_REMOTE_GREEN() {
+      mediaElement = test.pcRemote.remoteMediaElements[0];
+      ok(!!mediaElement, "Should have remote video element for pcRemote");
+      return h.pixelMustBecome(mediaElement, h.green, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become green",
+      });
+    },
+    function PC_LOCAL_DRAW_LOCAL_RED() {
+      // After requesting a frame it will be captured at the time of next render.
+      // Next render will happen at next stable state, at the earliest,
+      // i.e., this order of `requestFrame(); draw();` should work.
+      stream.requestFrame();
+      h.drawColor(canvas, h.red);
+    },
+    function PC_REMOTE_WAIT_FOR_REMOTE_RED() {
+      return h.pixelMustBecome(mediaElement, h.red, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become red",
+      });
+    }
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_2d_noSSRC.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_2d_noSSRC.html
new file mode 100644
index 0000000000..e33a7e8886
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_2d_noSSRC.html
@@ -0,0 +1,83 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "Canvas(2D)::CaptureStream as video-only input to peerconnection with no a=ssrc",
+  visible: true
+});
+
+var test;
+runNetworkTest(async (options) => {
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  options = options || { };
+  options.ssrc = false;
+  test = new PeerConnectionTest(options);
+  var mediaElement;
+  var h = new CaptureStreamTestHelper2D();
+  var canvas = document.createElement('canvas');
+  var stream;
+  canvas.id = 'source_canvas';
+  canvas.width = canvas.height = 16;
+  document.getElementById('content').appendChild(canvas);
+
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function PC_LOCAL_CANVAS_CAPTURESTREAM(test) {
+      h.drawColor(canvas, h.green);
+      stream = canvas.captureStream(0);
+      test.pcLocal.attachLocalStream(stream);
+      stream.requestFrame();
+      var i = 0;
+      return setInterval(function() {
+        try {
+          info("draw " + i ? "green" : "red");
+          h.drawColor(canvas, i ? h.green : h.red);
+          i = 1 - i;
+          stream.requestFrame();
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
+    }
+  ]);
+  test.chain.append([
+    function PC_REMOTE_WAIT_FOR_REMOTE_GREEN() {
+      mediaElement = test.pcRemote.remoteMediaElements[0];
+      ok(!!mediaElement, "Should have remote video element for pcRemote");
+      return h.pixelMustBecome(mediaElement, h.green, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become green",
+      });
+    },
+    function PC_LOCAL_DRAW_LOCAL_RED() {
+      // After requesting a frame it will be captured at the time of next render.
+      // Next render will happen at next stable state, at the earliest,
+      // i.e., this order of `requestFrame(); draw();` should work.
+      stream.requestFrame();
+      h.drawColor(canvas, h.red);
+    },
+    function PC_REMOTE_WAIT_FOR_REMOTE_RED() {
+      return h.pixelMustBecome(mediaElement, h.red, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become red",
+      });
+    }
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_webgl.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_webgl.html
new file mode 100644
index 0000000000..167379fb37
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_webgl.html
@@ -0,0 +1,130 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/webgl-mochitest/webgl-util.js"></script>
+</head>
+<body>
+<pre id="test">
+<script id="v-shader" type="x-shader/x-vertex">
+  attribute vec2 aPosition;
+  void main() {
+   gl_Position = vec4(aPosition, 0, 1);
+}
+</script>
+<script id="f-shader" type="x-shader/x-fragment">
+  precision mediump float;
+  uniform vec4 uColor;
+  void main() { gl_FragColor = uColor; }
+</script>
+<script type="application/javascript">
+createHTML({
+  bug: "1032848",
+  title: "Canvas(WebGL)::CaptureStream as video-only input to peerconnection"
+});
+
+runNetworkTest(async () => {
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  var test = new PeerConnectionTest();
+  var vremote;
+  var h = new CaptureStreamTestHelperWebGL();
+  var canvas = document.createElement('canvas');
+  canvas.id = 'source_canvas';
+  canvas.width = canvas.height = 16;
+  canvas.style.display = 'none';
+  document.getElementById('content').appendChild(canvas);
+
+  var gl = canvas.getContext('webgl');
+  if (!gl) {
+    todo(false, "WebGL unavailable.");
+    networkTestFinished();
+    return;
+  }
+
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function WEBGL_SETUP(test) {
+      var program = WebGLUtil.createProgramByIds(gl, 'v-shader', 'f-shader');
+
+      if (!program) {
+        ok(false, "Program should link");
+        return Promise.reject("Program should link");
+      }
+      gl.useProgram(program);
+
+      var uColorLocation = gl.getUniformLocation(program, "uColor");
+      h.setFragmentColorLocation(uColorLocation);
+
+      var squareBuffer = gl.createBuffer();
+      gl.bindBuffer(gl.ARRAY_BUFFER, squareBuffer);
+
+      var vertices = [ 0,  0,
+                      -1,  0,
+                       0,  1,
+                      -1,  1 ];
+      gl.bufferData(gl.ARRAY_BUFFER, new Float32Array(vertices), gl.STATIC_DRAW);
+      squareBuffer.itemSize = 2;
+      squareBuffer.numItems = 4;
+
+      program.aPosition = gl.getAttribLocation(program, "aPosition");
+      gl.enableVertexAttribArray(program.aPosition);
+      gl.vertexAttribPointer(program.aPosition, squareBuffer.itemSize, gl.FLOAT, false, 0, 0);
+    },
+    function PC_LOCAL_CANVAS_CAPTURESTREAM(test) {
+      h.drawColor(canvas, h.green);
+      test.pcLocal.canvasStream = canvas.captureStream(0.0);
+      is(test.pcLocal.canvasStream.canvas, canvas, "Canvas attribute is correct");
+      test.pcLocal.attachLocalStream(test.pcLocal.canvasStream);
+      var i = 0;
+      return setInterval(function() {
+        try {
+          info("draw " + i ? "green" : "red");
+          h.drawColor(canvas, i ? h.green : h.red);
+          i = 1 - i;
+          test.pcLocal.canvasStream.requestFrame();
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
+    }
+  ]);
+  test.chain.append([
+    function FIND_REMOTE_VIDEO() {
+      vremote = test.pcRemote.remoteMediaElements[0];
+      ok(!!vremote, "Should have remote video element for pcRemote");
+    },
+    function WAIT_FOR_REMOTE_GREEN() {
+      return h.pixelMustBecome(vremote, h.green, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become green",
+      });
+    },
+    function REQUEST_FRAME(test) {
+      // After requesting a frame it will be captured at the time of next render.
+      // Next render will happen at next stable state, at the earliest,
+      // i.e., this order of `requestFrame(); draw();` should work.
+      test.pcLocal.canvasStream.requestFrame();
+    },
+    function DRAW_LOCAL_RED() {
+      h.drawColor(canvas, h.red);
+    },
+    function WAIT_FOR_REMOTE_RED() {
+      return h.pixelMustBecome(vremote, h.red, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become red",
+      });
+    }
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_capturedVideo.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_capturedVideo.html
new file mode 100644
index 0000000000..f6f48ba429
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_capturedVideo.html
@@ -0,0 +1,81 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="pc.js"></script>
+  <script src="../../../test/manifest.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+(async () => {
+  await createHTML({
+    bug: "1081409",
+    title: "Captured video-only over peer connection",
+    visible: true
+  });
+
+  // Run tests in sequence for log readability.
+  PARALLEL_TESTS = 1;
+  const manager = new MediaTestManager;
+
+  async function startTest(media, token) {
+    manager.started(token);
+    info(`Starting test for ${media.name}`);
+    const video = document.createElement('video');
+    video.id = "id_" + media.name;
+    video.width = 160;
+    video.height = 120;
+    video.muted = true;
+    video.controls = true;
+    video.preload = "metadata";
+    video.src = "../../../test/" + media.name;
+
+    document.getElementById("content").appendChild(video);
+
+    const onerror = new Promise(r => video.onerror = r).then(_ =>
+      new Error(`${media.name} failed in playback. code=${video.error.code}`));
+
+    await Promise.race([
+      new Promise(res => video.onloadedmetadata = res),
+      onerror,
+    ]);
+    onerror.catch(e => ok(false, e));
+    setupEnvironment();
+    await testConfigured;
+    const stream = video.mozCaptureStream();
+    const test = new PeerConnectionTest(
+      {
+        config_local: { label_suffix: media.name },
+        config_remote: { label_suffix: media.name },
+      }
+    );
+    test.setOfferOptions(
+      {
+        offerToReceiveVideo: false,
+        offerToReceiveAudio: false,
+      }
+    );
+    const hasVideo = !!stream.getVideoTracks().length;
+    const hasAudio = !!stream.getAudioTracks().length;
+    test.setMediaConstraints([{ video: hasVideo, audio: hasAudio }], []);
+    test.chain.replace("PC_LOCAL_GUM", [
+      function PC_LOCAL_CAPTUREVIDEO(test) {
+        test.pcLocal.attachLocalStream(stream);
+      },
+    ]);
+    test.chain.insertBefore("PC_LOCAL_WAIT_FOR_MEDIA_FLOW", [
+      function PC_LOCAL_START_MEDIA(test) {
+        video.play();
+      },
+    ]);
+    await test.run();
+    removeNodeAndSource(video);
+    manager.finished(token);
+  }
+
+  manager.runTests(getPlayableVideos(gLongerTests), startTest);
+})();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_certificates.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_certificates.html
new file mode 100644
index 0000000000..561f285f60
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_certificates.html
@@ -0,0 +1,185 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1172785",
+    title: "Certificate management"
+  });
+
+  function badCertificate(config, expectedError, message) {
+    return RTCPeerConnection.generateCertificate(config)
+      .then(() => ok(false, message),
+            e => is(e.name, expectedError, message));
+  }
+
+  // Checks a handful of obviously bad options to RTCCertificate.create().  Most
+  // of the checking is done by the WebCrypto code underpinning this, hence the
+  // baffling error codes, but a sanity check is still in order.
+  function checkBadParameters() {
+    return Promise.all([
+      badCertificate({
+        name: "RSASSA-PKCS1-v1_5",
+        hash: "SHA-256",
+        modulusLength: 1023,
+        publicExponent: new Uint8Array([1, 0, 1])
+      }, "NotSupportedError", "1023-bit is too small to succeed"),
+
+      badCertificate({
+        name: "RSASSA-PKCS1-v1_5",
+          hash: "SHA-384",
+          modulusLength: 2048,
+          publicExponent: new Uint8Array([1, 0, 1])
+      }, "NotSupportedError", "SHA-384 isn't supported yet"),
+
+      // A SyntaxError happens in the "generate key operation" step, but
+      // webrtc-pc does not say to reject the promise if this step fails.
+      // It does say to throw NotSupportedError if we have passed "an
+      // algorithm that the user agent cannot or will not use to generate a
+      // certificate".
+      badCertificate({
+        name: "ECDH",
+        namedCurve: "P-256"
+      }, "NotSupportedError", "ECDH is rejected because the usage is neither \"deriveKey\" or \"deriveBits\""),
+
+      badCertificate({
+        name: "not a valid algorithm"
+      }, "NotSupportedError", "not a valid algorithm"),
+
+      badCertificate("ECDSA", "NotSupportedError", "a bare name is not enough"),
+
+      badCertificate({
+        name: "ECDSA",
+        namedCurve: "not a curve"
+      }, "NotSupportedError", "ECDSA with an unknown curve")
+    ]);
+  }
+
+  function createDB() {
+    var openDB = indexedDB.open("genericstore");
+    openDB.onupgradeneeded = e => {
+      var db = e.target.result;
+      db.createObjectStore("data");
+    };
+    return new Promise(resolve => {
+      openDB.onsuccess = e => resolve(e.target.result);
+    });
+  }
+
+  function resultPromise(tx, op) {
+    return new Promise((resolve, reject) => {
+      op.onsuccess = e => resolve(e.target.result);
+      op.onerror = () => reject(op.error);
+      tx.onabort = () => reject(tx.error);
+    });
+  }
+
+  function store(db, value) {
+    var tx = db.transaction("data", "readwrite");
+    var store = tx.objectStore("data");
+    return resultPromise(tx, store.put(value, "value"));
+  }
+
+  function retrieve(db) {
+    var tx = db.transaction("data", "readonly");
+    var store = tx.objectStore("data");
+    return resultPromise(tx, store.get("value"));
+  }
+
+  // Creates a database, stores a value, retrieves it.
+  function storeAndRetrieve(value) {
+    return createDB().then(db => {
+      return store(db, value)
+        .then(() => retrieve(db))
+        .then(retrieved => {
+          db.close();
+          return retrieved;
+        });
+    });
+  }
+
+  var test;
+  runNetworkTest(function (options) {
+    var expiredCert;
+    return Promise.resolve()
+      .then(() => RTCPeerConnection.generateCertificate({
+        name: "ECDSA",
+        namedCurve: "P-256",
+        expires: 1 // smallest possible expiration window
+      }))
+      .then(cert => {
+        ok(!isNaN(cert.expires), 'cert has expiration time');
+        info('Expires at ' + new Date(cert.expires));
+        expiredCert = cert;
+      })
+
+      .then(() => checkBadParameters())
+
+      .then(() => {
+        var delay = expiredCert.expires - Date.now();
+        // Hopefully this delay is never needed.
+        if (delay > 0) {
+          return new Promise(r => setTimeout(r, delay));
+        }
+      })
+      .then(() => {
+        ok(expiredCert.expires <= Date.now(), 'Cert should be at or past expiration');
+        try {
+          new RTCPeerConnection({ certificates: [expiredCert] });
+          ok(false, 'Constructing peer connection with an expired cert is not allowed');
+        } catch(e) {
+          is(e.name, 'InvalidAccessError',
+             'Constructing peer connection with an expired certs is not allowed');
+        }
+      })
+
+      .then(() => Promise.all([
+        RTCPeerConnection.generateCertificate({
+          name: "ECDSA",
+          namedCurve: "P-256"
+        }),
+        RTCPeerConnection.generateCertificate({
+          name: "RSASSA-PKCS1-v1_5",
+          hash: "SHA-256",
+          modulusLength: 2048,
+          publicExponent: new Uint8Array([1, 0, 1])
+        })
+      ]))
+
+    // A round trip through indexedDB should not do anything.
+      .then(storeAndRetrieve)
+      .then(certs => {
+        try {
+          new RTCPeerConnection({ certificates: certs });
+          ok(false, 'Constructing peer connection with multiple certs is not allowed');
+        } catch(e) {
+          is(e.name, 'NotSupportedError',
+             'Constructing peer connection with multiple certs is not allowed');
+        }
+        return certs;
+      })
+      .then(certs => {
+        test = new PeerConnectionTest({
+          config_local: {
+            certificates: [certs[0]]
+          },
+          config_remote: {
+            certificates: [certs[1]]
+          }
+        });
+        test.setMediaConstraints([{audio: true}], [{audio: true}]);
+        return test.run();
+      })
+      .catch(e => {
+        console.log('test failure', e);
+        ok(false, 'test failed: ' + e);
+      });
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_checkPacketDumpHook.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_checkPacketDumpHook.html
new file mode 100644
index 0000000000..248e102dd2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_checkPacketDumpHook.html
@@ -0,0 +1,107 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1377299",
+    title: "Check that packet dump hooks generate callbacks"
+  });
+
+  function waitForPacket(pc, checkFunction) {
+    return new Promise(resolve => {
+      function onPacket(level, type, sending, packet) {
+        if (checkFunction(level, type, sending, packet)) {
+          SpecialPowers.wrap(pc).mozSetPacketCallback(() => {});
+          resolve();
+        }
+      }
+
+      SpecialPowers.wrap(pc).mozSetPacketCallback(onPacket);
+    }
+    );
+  }
+
+  async function waitForSendPacket(pc, type, level) {
+    await SpecialPowers.wrap(pc).mozEnablePacketDump(level, type, true);
+    await timeout(
+      waitForPacket(pc, (obsLevel, obsType, sending) => {
+        is(obsLevel, level, "Level for packet is " + level);
+        is(obsType, type, "Type for packet is " + type);
+        ok(sending, "This is a send packet");
+        return true;
+      }),
+      10000, "Timeout waiting for " + type + " send packet on level " + level);
+    await SpecialPowers.wrap(pc).mozDisablePacketDump(level, type, true);
+  }
+
+  async function waitForRecvPacket(pc, type, level) {
+    await SpecialPowers.wrap(pc).mozEnablePacketDump(level, type, false);
+    await timeout(
+      waitForPacket(pc, (obsLevel, obsType, sending) => {
+        is(obsLevel, level, "Level for packet is " + level);
+        is(obsType, type, "Type for packet is " + type);
+        ok(!sending, "This is a recv packet");
+        return true;
+      }),
+      10000, "Timeout waiting for " + type + " recv packet on level " + level);
+    await SpecialPowers.wrap(pc).mozDisablePacketDump(level, type, false);
+  }
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true, video: true}],
+                             [{audio: true, video: true}]);
+    // pc.js uses video elements by default, we want to test audio elements here
+    test.pcLocal.audioElementsOnly = true;
+
+    test.chain.insertBefore('PC_LOCAL_WAIT_FOR_MEDIA_FLOW',[
+        async function PC_LOCAL_CHECK_PACKET_DUMP_HOOKS() {
+          await waitForRecvPacket(test.pcLocal._pc, "rtp", 0);
+          await waitForRecvPacket(test.pcLocal._pc, "rtcp", 0);
+          await waitForRecvPacket(test.pcLocal._pc, "srtp", 0);
+          await waitForRecvPacket(test.pcLocal._pc, "srtcp", 0);
+          await waitForSendPacket(test.pcLocal._pc, "rtp", 0);
+          await waitForSendPacket(test.pcLocal._pc, "rtcp", 0);
+          await waitForSendPacket(test.pcLocal._pc, "srtp", 0);
+          await waitForSendPacket(test.pcLocal._pc, "srtcp", 0);
+
+          await waitForRecvPacket(test.pcLocal._pc, "rtp", 1);
+          await waitForRecvPacket(test.pcLocal._pc, "rtcp", 1);
+          await waitForRecvPacket(test.pcLocal._pc, "srtp", 1);
+          await waitForRecvPacket(test.pcLocal._pc, "srtcp", 1);
+          await waitForSendPacket(test.pcLocal._pc, "rtp", 1);
+          await waitForSendPacket(test.pcLocal._pc, "rtcp", 1);
+          await waitForSendPacket(test.pcLocal._pc, "srtp", 1);
+          await waitForSendPacket(test.pcLocal._pc, "srtcp", 1);
+        },
+        async function PC_REMOTE_CHECK_PACKET_DUMP_HOOKS() {
+          await waitForRecvPacket(test.pcRemote._pc, "rtp", 0);
+          await waitForRecvPacket(test.pcRemote._pc, "rtcp", 0);
+          await waitForRecvPacket(test.pcRemote._pc, "srtp", 0);
+          await waitForRecvPacket(test.pcRemote._pc, "srtcp", 0);
+          await waitForSendPacket(test.pcRemote._pc, "rtp", 0);
+          await waitForSendPacket(test.pcRemote._pc, "rtcp", 0);
+          await waitForSendPacket(test.pcRemote._pc, "srtp", 0);
+          await waitForSendPacket(test.pcRemote._pc, "srtcp", 0);
+
+          await waitForRecvPacket(test.pcRemote._pc, "rtp", 1);
+          await waitForRecvPacket(test.pcRemote._pc, "rtcp", 1);
+          await waitForRecvPacket(test.pcRemote._pc, "srtp", 1);
+          await waitForRecvPacket(test.pcRemote._pc, "srtcp", 1);
+          await waitForSendPacket(test.pcRemote._pc, "rtp", 1);
+          await waitForSendPacket(test.pcRemote._pc, "rtcp", 1);
+          await waitForSendPacket(test.pcRemote._pc, "srtp", 1);
+          await waitForSendPacket(test.pcRemote._pc, "srtcp", 1);
+        }
+    ]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_close.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_close.html
new file mode 100644
index 0000000000..3edf677203
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_close.html
@@ -0,0 +1,134 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "991877",
+    title: "Basic RTCPeerConnection.close() tests"
+  });
+
+  runNetworkTest(function () {
+    var pc = new RTCPeerConnection();
+    var sender = pc.addTrack(getSilentTrack(), new MediaStream());
+    var exception = null;
+    var eTimeout = null;
+
+    // everything should be in initial state
+    is(pc.signalingState, "stable", "Initial signalingState is 'stable'");
+    is(pc.iceConnectionState, "new", "Initial iceConnectionState is 'new'");
+    is(pc.iceGatheringState, "new", "Initial iceGatheringState is 'new'");
+
+    var finish;
+    var finished = new Promise(resolve => finish = resolve);
+
+    var mustNotSettle = (p, ms, msg) => Promise.race([
+      p.then(() => ok(false, msg + " must not settle"),
+             e => ok(false, msg + " must not settle. Got " + e.name)),
+      wait(ms).then(() => ok(true, msg + " must not settle"))
+    ]);
+
+    var silence = mustNotSettle(pc.createOffer(), 1000,
+                                "createOffer immediately followed by close");
+    try {
+      pc.close();
+    } catch (e) {
+      exception = e;
+    }
+    is(exception, null, "closing the connection raises no exception");
+    is(pc.signalingState, "closed", "Final signalingState is 'closed'");
+    is(pc.iceConnectionState, "closed", "Final iceConnectionState is 'closed'");
+
+    // test that pc is really closed (and doesn't crash, bug 1259728)
+    try {
+      pc.getLocalStreams();
+    } catch (e) {
+      exception = e;
+    }
+    is(exception && exception.name, "InvalidStateError",
+       "pc.getLocalStreams should throw when closed");
+    exception = null;
+
+    try {
+      pc.close();
+    } catch (e) {
+      exception = e;
+    }
+    is(exception, null, "A second close() should not raise an exception");
+    is(pc.signalingState, "closed", "Final signalingState stays at 'closed'");
+    is(pc.iceConnectionState, "closed", "Final iceConnectionState stays at 'closed'");
+
+    // Due to a limitation in our WebIDL compiler that prevents overloads with
+    // both Promise and non-Promise return types, legacy APIs with callbacks
+    // are unable to continue to throw exceptions. Luckily the spec uses
+    // exceptions solely for "programming errors" so this should not hinder
+    // working code from working, which is the point of the legacy API. All
+    // new code should use the promise API.
+    //
+    // The legacy methods that no longer throw on programming errors like
+    // "invalid-on-close" are:
+    // - createOffer
+    // - createAnswer
+    // - setLocalDescription
+    // - setRemoteDescription
+    // - addIceCandidate
+    // - getStats
+    //
+    // These legacy methods fire the error callback instead. This is not
+    // entirely to spec but is better than ignoring programming errors.
+
+    var offer = new RTCSessionDescription({ sdp: "sdp", type: "offer" });
+    var answer = new RTCSessionDescription({ sdp: "sdp", type: "answer" });
+    var candidate = new RTCIceCandidate({ candidate: "dummy",
+                                             sdpMid: "test",
+                                             sdpMLineIndex: 3 });
+
+    var doesFail = (p, msg) => p.then(generateErrorCallback(msg),
+                                      r => is(r.name, "InvalidStateError", msg));
+    Promise.all([
+      [pc.createOffer(), "createOffer"],
+      [pc.createOffer({offerToReceiveAudio: true}), "createOffer({offerToReceiveAudio: true})"],
+      [pc.createOffer({offerToReceiveAudio: false}), "createOffer({offerToReceiveAudio: false})"],
+      [pc.createOffer({offerToReceiveVideo: true}), "createOffer({offerToReceiveVideo: true})"],
+      [pc.createOffer({offerToReceiveVideo: false}), "createOffer({offerToReceiveVideo: false})"],
+      [pc.createAnswer(), "createAnswer"],
+      [pc.setLocalDescription(offer), "setLocalDescription"],
+      [pc.setRemoteDescription(answer), "setRemoteDescription"],
+      [pc.addIceCandidate(candidate), "addIceCandidate"],
+      [new Promise((y, n) => pc.createOffer(y, n)), "Legacy createOffer"],
+      [new Promise((y, n) => pc.createAnswer(y, n)), "Legacy createAnswer"],
+      [new Promise((y, n) => pc.setLocalDescription(offer, y, n)), "Legacy setLocalDescription"],
+      [new Promise((y, n) => pc.setRemoteDescription(answer, y, n)), "Legacy setRemoteDescription"],
+      [new Promise((y, n) => pc.addIceCandidate(candidate, y, n)), "Legacy addIceCandidate"],
+      [sender.replaceTrack(getSilentTrack()), "replaceTrack"],
+    ].map(([p, name]) => doesFail(p, name + " fails on close")))
+    .catch(reason => ok(false, "unexpected failure: " + reason))
+    .then(finish);
+
+    // Other methods are unaffected.
+
+    SimpleTest.doesThrow(function() {
+      pc.updateIce("Invalid RTC Configuration")},
+      "updateIce() on closed PC raised expected exception");
+
+    SimpleTest.doesThrow(function() {
+      pc.addStream("Invalid Media Stream")},
+      "addStream() on closed PC raised expected exception");
+
+    SimpleTest.doesThrow(function() {
+      pc.createDataChannel({})},
+      "createDataChannel() on closed PC raised expected exception");
+
+    SimpleTest.doesThrow(function() {
+      pc.setIdentityProvider("Invalid Provider")},
+      "setIdentityProvider() on closed PC raised expected exception");
+
+    return Promise.all([finished, silence]);
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_closeDuringIce.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_closeDuringIce.html
new file mode 100644
index 0000000000..db3a2922d5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_closeDuringIce.html
@@ -0,0 +1,79 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1087629",
+    title: "Close PCs during ICE connectivity check"
+  });
+
+// Test closeDuringIce to simulate problems during peer connections
+
+
+function PC_LOCAL_SETUP_NULL_ICE_HANDLER(test) {
+  test.pcLocal.setupIceCandidateHandler(test, function() {}, function () {});
+}
+function PC_REMOTE_SETUP_NULL_ICE_HANDLER(test) {
+  test.pcRemote.setupIceCandidateHandler(test, function() {}, function () {});
+}
+function PC_REMOTE_ADD_FAKE_ICE_CANDIDATE(test) {
+  var cand = {"candidate":"candidate:0 1 UDP 2130379007 192.0.2.1 12345 typ host","sdpMid":"","sdpMLineIndex":0};
+  test.pcRemote.storeOrAddIceCandidate(cand);
+  info(test.pcRemote + " Stored fake candidate: " + JSON.stringify(cand));
+}
+function PC_LOCAL_ADD_FAKE_ICE_CANDIDATE(test) {
+  var cand = {"candidate":"candidate:0 1 UDP 2130379007 192.0.2.2 56789 typ host","sdpMid":"","sdpMLineIndex":0};
+    test.pcLocal.storeOrAddIceCandidate(cand);
+  info(test.pcLocal + " Stored fake candidate: " + JSON.stringify(cand));
+}
+function PC_LOCAL_CLOSE_DURING_ICE(test) {
+  return test.pcLocal.iceChecking.then(() => {
+    test.pcLocal.onsignalingstatechange = function () {};
+    test.pcLocal.close();
+    });
+}
+function PC_REMOTE_CLOSE_DURING_ICE(test) {
+  return test.pcRemote.iceChecking.then(() => {
+    test.pcRemote.onsignalingstatechange = function () {};
+    test.pcRemote.close();
+    });
+}
+function PC_LOCAL_WAIT_FOR_ICE_CHECKING(test) {
+  var resolveIceChecking;
+  test.pcLocal.iceChecking = new Promise(r => resolveIceChecking = r);
+  test.pcLocal.ice_connection_callbacks.checkIceStatus = () => {
+	if (test.pcLocal._pc.iceConnectionState === "checking") {
+	  resolveIceChecking();
+	}
+  }
+}
+function PC_REMOTE_WAIT_FOR_ICE_CHECKING(test) {
+  var resolveIceChecking;
+  test.pcRemote.iceChecking = new Promise(r => resolveIceChecking = r);
+  test.pcRemote.ice_connection_callbacks.checkIceStatus = () => {
+	if (test.pcRemote._pc.iceConnectionState === "checking") {
+	  resolveIceChecking();
+	}
+  }
+}
+
+runNetworkTest(() => {
+  var test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.replace("PC_LOCAL_SETUP_ICE_HANDLER", PC_LOCAL_SETUP_NULL_ICE_HANDLER);
+  test.chain.replace("PC_REMOTE_SETUP_ICE_HANDLER", PC_REMOTE_SETUP_NULL_ICE_HANDLER);
+  test.chain.insertAfter("PC_REMOTE_SETUP_NULL_ICE_HANDLER", PC_LOCAL_WAIT_FOR_ICE_CHECKING);
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_ICE_CHECKING", PC_REMOTE_WAIT_FOR_ICE_CHECKING);
+  test.chain.removeAfter("PC_LOCAL_SET_REMOTE_DESCRIPTION");
+  test.chain.append([PC_REMOTE_ADD_FAKE_ICE_CANDIDATE, PC_LOCAL_ADD_FAKE_ICE_CANDIDATE,
+	PC_LOCAL_CLOSE_DURING_ICE, PC_REMOTE_CLOSE_DURING_ICE]);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_codecNegotiationFailure.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_codecNegotiationFailure.html
new file mode 100644
index 0000000000..819e13fe1b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_codecNegotiationFailure.html
@@ -0,0 +1,111 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="iceTestUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1683934",
+    title: "RTCPeerConnection check codec negotiation failure"
+  });
+
+  function makeWeirdCodecs(sdp) {
+    return sdp
+      .replaceAll('VP8', 'VEEEEEEEEP8')
+      .replaceAll('VP9', 'VEEEEEEEEP9')
+      .replaceAll('H264', 'HERP264');
+  }
+
+  const tests = [
+    async function offererWeirdCodecs() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      const offer = await pc1.createOffer();
+      offer.sdp = makeWeirdCodecs(offer.sdp);
+      // It is not an error to receive an offer with no codecs we support
+      await pc2.setRemoteDescription(offer);
+      await pc2.setLocalDescription();
+      await wait(2000);
+    },
+
+    async function answererWeirdCodecs() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await pc1.setLocalDescription();
+      await pc2.setRemoteDescription(pc1.localDescription);
+      const answer = await pc2.createAnswer();
+      answer.sdp = makeWeirdCodecs(answer.sdp);
+      try {
+        await pc1.setRemoteDescription(answer);
+        ok(false, "Should have thrown");
+      } catch (e) {
+        ok(true, "Should have thrown");
+      }
+    },
+
+    async function reoffererWeirdCodecs() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      const offer = await pc1.createOffer();
+      offer.sdp = makeWeirdCodecs(offer.sdp);
+      // It is not an error to receive an offer with no codecs we support
+      await pc2.setRemoteDescription(offer);
+      await pc2.setLocalDescription();
+      await wait(2000);
+    },
+
+    async function reanswererWeirdCodecs() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+      await pc1.setLocalDescription();
+      await pc2.setRemoteDescription(pc1.localDescription);
+      const answer = await pc2.createAnswer();
+      answer.sdp = makeWeirdCodecs(answer.sdp);
+      try {
+        await pc1.setRemoteDescription(answer);
+        ok(false, "Should have thrown");
+      } catch (e) {
+        ok(true, "Should have thrown");
+      }
+    },
+
+  ];
+
+  runNetworkTest(async () => {
+    for (const test of tests) {
+      info(`Running test: ${test.name}`);
+      await test();
+      info(`Done running test: ${test.name}`);
+    }
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_constructedStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_constructedStream.html
new file mode 100644
index 0000000000..8431b7534e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_constructedStream.html
@@ -0,0 +1,67 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1271669",
+  title: "Test that pc.addTrack() accepts any MediaStream",
+  visible: true
+});
+
+runNetworkTest(() => {
+  var test = new PeerConnectionTest();
+  var constructedStream;
+  var dummyStream = new MediaStream();
+  var dummyStreamTracks = [];
+
+  test.setMediaConstraints([ {audio: true, video: true}
+                           , {audio: true}
+                           , {video: true}
+                           ], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function PC_LOCAL_GUM_CONSTRUCTED_STREAM(test) {
+      return getUserMedia(test.pcLocal.constraints[0]).then(stream => {
+        constructedStream = new MediaStream(stream.getTracks());
+        test.pcLocal.attachLocalStream(constructedStream);
+      });
+    },
+    function PC_LOCAL_GUM_DUMMY_STREAM(test) {
+      return getUserMedia(test.pcLocal.constraints[1])
+        .then(stream => dummyStreamTracks.push(...stream.getTracks()))
+        .then(() => getUserMedia(test.pcLocal.constraints[2]))
+        .then(stream => dummyStreamTracks.push(...stream.getTracks()))
+        .then(() => dummyStreamTracks.forEach(t =>
+          test.pcLocal.attachLocalTrack(t, dummyStream)));
+    },
+  ]);
+
+  let checkSentTracksReceived = (sentStreamId, sentTracks) => {
+    let receivedStream =
+      test.pcRemote._pc.getRemoteStreams().find(s => s.id == sentStreamId);
+    ok(receivedStream, "We should receive a stream with with the sent stream's id (" + sentStreamId + ")");
+    if (!receivedStream) {
+      return;
+    }
+
+    is(receivedStream.getTracks().length, sentTracks.length,
+       "Should receive same number of tracks as were sent");
+  };
+
+  test.chain.append([
+    function PC_REMOTE_CHECK_RECEIVED_CONSTRUCTED_STREAM() {
+      checkSentTracksReceived(constructedStream.id, constructedStream.getTracks());
+    },
+    function PC_REMOTE_CHECK_RECEIVED_DUMMY_STREAM() {
+      checkSentTracksReceived(dummyStream.id, dummyStreamTracks);
+    },
+  ]);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_disabledVideoPreNegotiation.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_disabledVideoPreNegotiation.html
new file mode 100644
index 0000000000..4c06de792e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_disabledVideoPreNegotiation.html
@@ -0,0 +1,45 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1570673",
+    title: "Sending an initially disabled video track should be playable remotely",
+    visible: true,
+  });
+
+  var test;
+  runNetworkTest(async (options) => {
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], []);
+    test.chain.insertAfter("PC_LOCAL_GUM", function PC_LOCAL_DISABLE_VIDEO() {
+      for (const {track} of test.pcLocal._pc.getSenders()) {
+        if (track.kind == "video") {
+          track.enabled = false;
+        }
+      }
+    });
+    test.chain.append(async function PC_REMOTE_RECEIVING_BLACK() {
+      const v = test.pcRemote.remoteMediaElements[0];
+      is(v.readyState, v.HAVE_ENOUGH_DATA, "video element should be playing");
+      const h = new CaptureStreamTestHelper2D();
+      await h.waitForPixel(test.pcRemote.remoteMediaElements[0],
+        px => h.isPixel(px, h.black, 128));
+    });
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_encodingsNegotiation.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_encodingsNegotiation.html
new file mode 100644
index 0000000000..f46d7eb0d2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_encodingsNegotiation.html
@@ -0,0 +1,85 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1401592",
+  title: "Simulcast negotiation tests",
+  visible: true
+});
+
+// simulcast negotiation is mostly tested in wpt, but we test a few
+// implementation-specific things here.
+const tests = [
+  async function checkVideoEncodingLimit() {
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    pc2.addTrack(stream.getTracks()[0]);
+
+    await doOfferToRecvSimulcast(pc2, pc1, ["1", "2", "3", "4"]);
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["1", "2", "3"]);
+
+    pc1.close();
+    pc2.close();
+    stream.getTracks().forEach(track => track.stop());
+  },
+
+  // wpt currently does not assume support for 3 encodings, which limits the
+  // effectiveness of its powers-of-2 test (since it can test only for 1 and 2)
+  async function checkScaleResolutionDownByAutoFillPowersOf2() {
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    pc2.addTrack(stream.getTracks()[0]);
+
+    await doOfferToRecvSimulcast(pc2, pc1, ["1", "2", "3"]);
+
+    const {encodings} = sender.getParameters();
+    const scaleValues = encodings.map(({scaleResolutionDownBy}) => scaleResolutionDownBy);
+    isDeeply(scaleValues, [4, 2, 1]);
+  },
+
+  async function checkLibwebrtcRidLengthLimit() {
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    pc2.addTrack(stream.getTracks()[0]);
+
+    await doOfferToRecvSimulcast(pc2, pc1, ["foo", "wibblywobblyjeremybearimy"]);
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["foo"]);
+
+    pc1.close();
+    pc2.close();
+    stream.getTracks().forEach(track => track.stop());
+  },
+];
+
+runNetworkTest(async () => {
+  for (const test of tests) {
+    info(`Running test: ${test.name}`);
+    await test();
+    info(`Done running test: ${test.name}`);
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_errorCallbacks.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_errorCallbacks.html
new file mode 100644
index 0000000000..851a256509
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_errorCallbacks.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "834270",
+    title: "Align PeerConnection error handling with WebRTC specification"
+  });
+
+  function validateReason(reason) {
+    ok(reason.name.length, "Reason name = " + reason.name);
+    ok(reason.message.length, "Reason message = " + reason.message);
+  };
+
+  function testCreateAnswerError() {
+    var pc = new RTCPeerConnection();
+    info ("Testing createAnswer error");
+    return pc.createAnswer()
+    .then(generateErrorCallback("createAnswer before offer should fail"),
+          validateReason);
+  };
+
+  function testSetLocalDescriptionError() {
+    var pc = new RTCPeerConnection();
+    info ("Testing setLocalDescription error");
+    return pc.setLocalDescription({ sdp: "Picklechips!", type: "offer" })
+    .then(generateErrorCallback("setLocalDescription with nonsense SDP should fail"),
+          validateReason);
+  };
+
+  function testSetRemoteDescriptionError() {
+    var pc = new RTCPeerConnection();
+    info ("Testing setRemoteDescription error");
+    return pc.setRemoteDescription({ sdp: "Who?", type: "offer" })
+    .then(generateErrorCallback("setRemoteDescription with nonsense SDP should fail"),
+          validateReason);
+  };
+
+  // No test for createOffer errors -- there's nothing we can do at this
+  // level to evoke an error in createOffer.
+
+  runNetworkTest(function () {
+    return testCreateAnswerError()
+    .then(testSetLocalDescriptionError)
+    .then(testSetRemoteDescriptionError);
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_extmapRenegotiation.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_extmapRenegotiation.html
new file mode 100644
index 0000000000..78c6bb986c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_extmapRenegotiation.html
@@ -0,0 +1,325 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="iceTestUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1799932",
+    title: "RTCPeerConnection check renegotiation of extmap"
+  });
+
+  function setExtmap(sdp, uri, id) {
+    const regex = new RegExp(`a=extmap:[0-9]+(\/[a-z]+)? ${uri}`, 'g');
+    if (id) {
+      return sdp.replaceAll(regex, `a=extmap:${id}$1 ${uri}`);
+    } else {
+      return sdp.replaceAll(regex, `a=unknownattr`);
+    }
+  }
+
+  function getExtmap(sdp, uri) {
+    const regex = new RegExp(`a=extmap:([0-9]+)(\/[a-z]+)? ${uri}`);
+    return sdp.match(regex)[1];
+  }
+
+  function replaceExtUri(sdp, oldUri, newUri) {
+    const regex = new RegExp(`(a=extmap:[0-9]+\/[a-z]+)? ${oldUri}`, 'g');
+    return sdp.replaceAll(regex, `$1 ${newUri}`);
+  }
+
+  const tests = [
+    async function checkAudioMidChange() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      // Sadly, there's no way to tell the offerer to change the extmap. Other
+      // types of endpoint could conceivably do this, so we at least don't want
+      // to crash.
+      // TODO: Would be nice to be able to test this with an endpoint that
+      // actually changes the ids it uses.
+      const reoffer = await pc1.createOffer();
+      reoffer.sdp = setExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", 14);
+      info(`New reoffer: ${reoffer.sdp}`);
+      await pc2.setRemoteDescription(reoffer);
+      await pc2.setLocalDescription();
+      await wait(2000);
+    },
+
+    async function checkVideoMidChange() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      // Sadly, there's no way to tell the offerer to change the extmap. Other
+      // types of endpoint could conceivably do this, so we at least don't want
+      // to crash.
+      // TODO: Would be nice to be able to test this with an endpoint that
+      // actually changes the ids it uses.
+      const reoffer = await pc1.createOffer();
+      reoffer.sdp = setExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", 14);
+      info(`New reoffer: ${reoffer.sdp}`);
+      await pc2.setRemoteDescription(reoffer);
+      await pc2.setLocalDescription();
+      await wait(2000);
+    },
+
+    async function checkAudioMidSwap() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      // Sadly, there's no way to tell the offerer to change the extmap. Other
+      // types of endpoint could conceivably do this, so we at least don't want
+      // to crash.
+      // TODO: Would be nice to be able to test this with an endpoint that
+      // actually changes the ids it uses.
+      const reoffer = await pc1.createOffer();
+      const midId = getExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid");
+      const ssrcLevelId = getExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level");
+      reoffer.sdp = setExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", ssrcLevelId);
+      reoffer.sdp = setExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", midId);
+      info(`New reoffer: ${reoffer.sdp}`);
+      try {
+        await pc2.setRemoteDescription(reoffer);
+        ok(false, "sRD should fail when it attempts extension id remapping");
+      } catch (e) {
+        ok(true, "sRD should fail when it attempts extension id remapping");
+      }
+    },
+
+    async function checkVideoMidSwap() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      // Sadly, there's no way to tell the offerer to change the extmap. Other
+      // types of endpoint could conceivably do this, so we at least don't want
+      // to crash.
+      // TODO: Would be nice to be able to test this with an endpoint that
+      // actually changes the ids it uses.
+      const reoffer = await pc1.createOffer();
+      const midId = getExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid");
+      const toffsetId = getExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:toffset");
+      reoffer.sdp = setExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", toffsetId);
+      reoffer.sdp = setExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:toffset", midId);
+      info(`New reoffer: ${reoffer.sdp}`);
+      try {
+        await pc2.setRemoteDescription(reoffer);
+        ok(false, "sRD should fail when it attempts extension id remapping");
+      } catch (e) {
+        ok(true, "sRD should fail when it attempts extension id remapping");
+      }
+    },
+
+    async function checkAudioIdReuse() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      // Sadly, there's no way to tell the offerer to change the extmap. Other
+      // types of endpoint could conceivably do this, so we at least don't want
+      // to crash.
+      // TODO: Would be nice to be able to test this with an endpoint that
+      // actually changes the ids it uses.
+      const reoffer = await pc1.createOffer();
+      // Change uri, but not the id, so the id now refers to foo.
+      reoffer.sdp = replaceExtUri(reoffer.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", "foo");
+      info(`New reoffer: ${reoffer.sdp}`);
+      try {
+        await pc2.setRemoteDescription(reoffer);
+        ok(false, "sRD should fail when it attempts extension id remapping");
+      } catch (e) {
+        ok(true, "sRD should fail when it attempts extension id remapping");
+      }
+    },
+
+    async function checkVideoIdReuse() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      // Sadly, there's no way to tell the offerer to change the extmap. Other
+      // types of endpoint could conceivably do this, so we at least don't want
+      // to crash.
+      // TODO: Would be nice to be able to test this with an endpoint that
+      // actually changes the ids it uses.
+      const reoffer = await pc1.createOffer();
+      // Change uri, but not the id, so the id now refers to foo.
+      reoffer.sdp = replaceExtUri(reoffer.sdp, "urn:ietf:params:rtp-hdrext:toffset", "foo");
+      info(`New reoffer: ${reoffer.sdp}`);
+      try {
+        await pc2.setRemoteDescription(reoffer);
+        ok(false, "sRD should fail when it attempts extension id remapping");
+      } catch (e) {
+        ok(true, "sRD should fail when it attempts extension id remapping");
+      }
+    },
+
+    // What happens when remote answer uses an extmap id, and then a remote
+    // reoffer tries to use the same id for something else?
+    async function checkAudioIdReuseOffererThenAnswerer() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      const reoffer = await pc2.createOffer();
+      // Change uri, but not the id, so the id now refers to foo.
+      reoffer.sdp = replaceExtUri(reoffer.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", "foo");
+      info(`New reoffer: ${reoffer.sdp}`);
+      try {
+        await pc1.setRemoteDescription(reoffer);
+        ok(false, "sRD should fail when it attempts extension id remapping");
+      } catch (e) {
+        ok(true, "sRD should fail when it attempts extension id remapping");
+      }
+    },
+
+    // What happens when a remote offer uses a different extmap id than the
+    // default? Does the answerer remember the new id in reoffers?
+    async function checkAudioIdReuseOffererThenAnswerer() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      // Negotiate, but change id for ssrc-audio-level to something pc2 would
+      // not typically use.
+      await pc1.setLocalDescription();
+      const mungedOffer = setExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", 12);
+      await pc2.setRemoteDescription({sdp: mungedOffer, type: "offer"});
+      await pc2.setLocalDescription();
+
+      const reoffer = await pc2.createOffer();
+      is(getExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level"), "12");
+    },
+
+    async function checkAudioUnnegotiatedIdReuse1() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      // Negotiate, but remove ssrc-audio-level from answer
+      await pc1.setLocalDescription();
+      const levelId = getExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level");
+      await pc2.setRemoteDescription(pc1.localDescription);
+      await pc2.setLocalDescription();
+      const answerNoExt = setExtmap(pc2.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", undefined);
+      await pc1.setRemoteDescription({sdp: answerNoExt, type: "answer"});
+
+      // Renegotiate, and use the id that offerer used for ssrc-audio-level for
+      // something different (while making sure we don't use it twice)
+      await pc2.setLocalDescription();
+      const mungedReoffer = setExtmap(pc2.localDescription.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", levelId);
+      const twiceMungedReoffer = setExtmap(mungedReoffer, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", undefined);
+      await pc1.setRemoteDescription({sdp: twiceMungedReoffer, type: "offer"});
+    },
+
+    async function checkAudioUnnegotiatedIdReuse2() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      // Negotiate, but remove ssrc-audio-level from offer. pc2 has never seen
+      // |levelId| in extmap yet, but internally probably wants to use that for
+      // ssrc-audio-level
+      await pc1.setLocalDescription();
+      const levelId = getExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level");
+      const offerNoExt = setExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", undefined);
+      await pc2.setRemoteDescription({sdp: offerNoExt, type: "offer"});
+      await pc2.setLocalDescription();
+      await pc1.setRemoteDescription(pc2.localDescription);
+
+      // Renegotiate, but use |levelId| for something other than
+      // ssrc-audio-level. pc2 should not throw.
+      await pc1.setLocalDescription();
+      const mungedReoffer = setExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", levelId);
+      const twiceMungedReoffer = setExtmap(mungedReoffer, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", undefined);
+      await pc2.setRemoteDescription({sdp: twiceMungedReoffer, type: "offer"});
+    },
+
+    async function checkAudioUnnegotiatedIdReuse3() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      // Negotiate, but replace ssrc-audio-level with something pc2 won't
+      // support in offer.
+      await pc1.setLocalDescription();
+      const levelId = getExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level");
+      const mungedOffer = replaceExtUri(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", "fooba");
+      await pc2.setRemoteDescription({sdp: mungedOffer, type: "offer"});
+      await pc2.setLocalDescription();
+      await pc1.setRemoteDescription(pc2.localDescription);
+
+      // Renegotiate, and use levelId for something pc2 _will_ support.
+      await pc1.setLocalDescription();
+      const mungedReoffer = setExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", levelId);
+      const twiceMungedReoffer = setExtmap(mungedReoffer, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", undefined);
+      await pc2.setRemoteDescription({sdp: twiceMungedReoffer, type: "offer"});
+    },
+
+  ];
+
+  runNetworkTest(async () => {
+    for (const test of tests) {
+      info(`Running test: ${test.name}`);
+      await test();
+      info(`Done running test: ${test.name}`);
+    }
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_forwarding_basicAudioVideoCombined.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_forwarding_basicAudioVideoCombined.html
new file mode 100644
index 0000000000..84b53a123b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_forwarding_basicAudioVideoCombined.html
@@ -0,0 +1,41 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "931903",
+    title: "Forwarding a stream from a combined audio/video peerconnection to another"
+  });
+
+runNetworkTest(function() {
+  var gumTest = new PeerConnectionTest();
+
+  var forwardingOptions = { config_local: { label_suffix: "forwarded" },
+                            config_remote: { label_suffix: "forwarded" } };
+  var forwardingTest = new PeerConnectionTest(forwardingOptions);
+
+  gumTest.setMediaConstraints([{audio: true, video: true}], []);
+  forwardingTest.setMediaConstraints([{audio: true, video: true}], []);
+  forwardingTest.chain.replace("PC_LOCAL_GUM", [
+    function PC_FORWARDING_CAPTUREVIDEO(test) {
+      var streams = gumTest.pcRemote._pc.getRemoteStreams();
+      is(streams.length, 1, "One stream to forward");
+      is(streams[0].getTracks().length, 2, "Forwarded stream has 2 tracks");
+      forwardingTest.pcLocal.attachLocalStream(streams[0]);
+      return Promise.resolve();
+    }
+  ]);
+  gumTest.chain.removeAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW");
+  return gumTest.chain.execute()
+    .then(() => forwardingTest.chain.execute())
+    .then(() => gumTest.close())
+    .then(() => forwardingTest.close());
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithSetConfiguration.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithSetConfiguration.html
new file mode 100644
index 0000000000..6710e628aa
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithSetConfiguration.html
@@ -0,0 +1,450 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="iceTestUtils.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script></head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1253706",
+    title: "Test ICE gathering when setConfiguration is used to change the ICE config"
+  });
+
+  const tests = [
+    async function baselineV4Cases() {
+      await checkSrflx([{urls:[`stun:${turnAddressV4}`]}]);
+      await checkRelayUdp([{urls:[`turn:${turnAddressV4}`], username, credential}]);
+      await checkRelayTcp([{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]);
+      await checkRelayUdpTcp([{urls:[`turn:${turnAddressV4}`, `turn:${turnAddressV4}?transport=tcp`], username, credential}]);
+      await checkNoSrflx();
+      await checkNoRelay();
+    },
+
+    async function addStunServerBeforeOffer() {
+      const pc = new RTCPeerConnection();
+      try {
+        pc.setConfiguration({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+        const candidates = await gatherWithTimeout(pc, 32000, `just a stun server`);
+        ok(candidates.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(!candidates.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addTurnServerBeforeOffer() {
+      const pc = new RTCPeerConnection();
+      try {
+        pc.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+        const candidates = await gatherWithTimeout(pc, 32000, `a turn (udp) server`);
+        ok(candidates.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(candidates.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addTurnTcpServerBeforeOffer() {
+      const pc = new RTCPeerConnection();
+      try {
+        pc.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]});
+        const candidates = await gatherWithTimeout(pc, 32000, `a turn (tcp) server`);
+        ok(!candidates.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(candidates.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addStunServerAfterOffer() {
+      const pc = new RTCPeerConnection();
+      try {
+        const candidates1 = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates1.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates1.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+        await pc.setLocalDescription({type: "rollback"});
+
+        pc.setConfiguration({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+        const candidates2 = await gatherWithTimeout(pc, 32000, `just a stun server`);
+        ok(candidates2.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(!candidates2.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addTurnServerAfterOffer() {
+      const pc = new RTCPeerConnection();
+      try {
+        const candidates1 = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates1.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates1.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+        await pc.setLocalDescription({type: "rollback"});
+
+        pc.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+        const candidates2 = await gatherWithTimeout(pc, 32000, `a turn (udp) server`);
+        ok(candidates2.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(candidates2.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addTurnTcpServerAfterOffer() {
+      const pc = new RTCPeerConnection();
+      try {
+        const candidates1 = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates1.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates1.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+        await pc.setLocalDescription({type: "rollback"});
+
+        pc.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]});
+        const candidates2 = await gatherWithTimeout(pc, 32000, `a turn (tcp) server`);
+        ok(!candidates2.some(c => c.candidate.includes("srflx")), "Should get no srflx candidates");
+        ok(candidates2.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function removeStunServerBeforeOffer() {
+      const pc = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+      try {
+        pc.setConfiguration({});
+        const candidates = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function removeTurnServerBeforeOffer() {
+      const pc = new RTCPeerConnection({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+      try {
+        pc.setConfiguration({});
+        const candidates = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function removeTurnTcpServerBeforeOffer() {
+      const pc = new RTCPeerConnection({iceServers: [{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]});
+      try {
+        pc.setConfiguration({});
+        const candidates = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function removeStunServerAfterOffer() {
+      const pc = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+      try {
+        const candidates1 = await gatherWithTimeout(pc, 32000, `just a stun server`);
+        ok(candidates1.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(!candidates1.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+        await pc.setLocalDescription({type: "rollback"});
+
+        pc.setConfiguration({});
+        const candidates2 = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates2.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates2.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function removeTurnServerAfterOffer() {
+      const pc = new RTCPeerConnection({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+      try {
+        const candidates1 = await gatherWithTimeout(pc, 32000, `a turn (udp) server`);
+        ok(candidates1.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(candidates1.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+        await pc.setLocalDescription({type: "rollback"});
+
+        pc.setConfiguration({});
+        const candidates2 = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates2.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates2.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function removeTurnTcpServerAfterOffer() {
+      const pc = new RTCPeerConnection({iceServers: [{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]});
+      try {
+        const candidates1 = await gatherWithTimeout(pc, 32000, `a turn (tcp) server`);
+        ok(!candidates1.some(c => c.candidate.includes("srflx")), "Should get no srflx candidates");
+        ok(candidates1.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+        await pc.setLocalDescription({type: "rollback"});
+
+        pc.setConfiguration({});
+        const candidates2 = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates2.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates2.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addStunServerAfterNegotiation() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+      try {
+        const candidatePromise = trickleIce(offerer);
+        await connect(offerer, answerer, 32000, `no ICE servers`);
+        const candidates = await candidatePromise;
+        const ufrags = Array.from(new Set(candidates.map(c => c.usernameFragment)));
+        is(ufrags.length, 1, "Should have one ufrag in candidate set");
+
+        offerer.setConfiguration({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+        const candidates2 = await gatherWithTimeout(offerer, 32000, `just a stun server`);
+        ok(candidates2.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(!candidates2.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+        const ufrags2 = Array.from(new Set(candidates2.map(c => c.usernameFragment)));
+        is(ufrags2.length, 1, "Should have one ufrag in candidate set");
+        isnot(ufrags[0], ufrags2[0], "ufrag should change, because setConfiguration should have triggered an ICE restart");
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addTurnServerAfterNegotiation() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+      try {
+        const candidatePromise = trickleIce(offerer);
+        await connect(offerer, answerer, 32000, `no ICE servers`);
+        const candidates = await candidatePromise;
+        const ufrags = Array.from(new Set(candidates.map(c => c.usernameFragment)));
+        is(ufrags.length, 1, "Should have one ufrag in candidate set");
+
+        offerer.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+        const candidates2 = await gatherWithTimeout(offerer, 32000, `a turn (udp) server`);
+        ok(candidates2.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(candidates2.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+        const ufrags2 = Array.from(new Set(candidates2.map(c => c.usernameFragment)));
+        is(ufrags2.length, 1, "Should have one ufrag in candidate set");
+        isnot(ufrags[0], ufrags2[0], "ufrag should change, because setConfiguration should have triggered an ICE restart");
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addTurnTcpServerAfterNegotiation() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+      try {
+        const candidatePromise = trickleIce(offerer);
+        await connect(offerer, answerer, 32000, `no ICE servers`);
+        const candidates = await candidatePromise;
+        const ufrags = Array.from(new Set(candidates.map(c => c.usernameFragment)));
+        is(ufrags.length, 1, "Should have one ufrag in candidate set");
+
+        offerer.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]});
+        const candidates2 = await gatherWithTimeout(offerer, 32000, `a turn (tcp) server`);
+        ok(!candidates2.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(candidates2.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+        const ufrags2 = Array.from(new Set(candidates2.map(c => c.usernameFragment)));
+        is(ufrags2.length, 1, "Should have one ufrag in candidate set");
+        isnot(ufrags[0], ufrags2[0], "ufrag should change, because setConfiguration should have triggered an ICE restart");
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addStunServerBeforeCreateAnswer() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection();
+
+      try {
+        await answerer.setRemoteDescription(await offerer.createOffer({offerToReceiveAudio: true}));
+
+        answerer.setConfiguration({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+        const candidates = await gatherWithTimeout(answerer, 32000, `just a stun server`);
+        ok(candidates.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(!candidates.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addTurnServerBeforeCreateAnswer() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection();
+
+      try {
+        await answerer.setRemoteDescription(await offerer.createOffer({offerToReceiveAudio: true}));
+
+        answerer.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+        const candidates = await gatherWithTimeout(answerer, 32000, `a turn (udp) server`);
+        ok(candidates.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(candidates.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addTurnTcpServerBeforeCreateAnswer() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection();
+
+      try {
+        await answerer.setRemoteDescription(await offerer.createOffer({offerToReceiveAudio: true}));
+
+        answerer.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]});
+        const candidates = await gatherWithTimeout(answerer, 32000, `a turn (tcp) server`);
+        ok(!candidates.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(candidates.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function relayPolicyPreventsSrflx() {
+      const pc = new RTCPeerConnection();
+      try {
+        pc.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}], iceTransportPolicy: "relay"});
+        const candidates = await gatherWithTimeout(pc, 32000, `a turn (udp) server`);
+        ok(!candidates.some(c => c.candidate.includes("srflx")), "Should not get a srflx candidate");
+        ok(candidates.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addOffererStunServerAllowsIceToConnect() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection();
+
+      try {
+        try {
+          // Both ends are behind a simulated endpoint-independent NAT, which
+          // requires at least one side to have a srflx candidate to work.
+          await connect(offerer, answerer, 2000, `no ICE servers`);
+          ok(false, "ICE should either have failed, or timed out!");
+        } catch (e) {
+          if (!(e instanceof Error)) throw e;
+          ok(true, "ICE should either have failed, or timed out!");
+        }
+
+        offerer.setConfiguration({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+        await connect(offerer, answerer, 32000, `just a STUN server`);
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addAnswererStunServerDoesNotAllowIceToConnect() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection();
+
+      try {
+        try {
+          // Both ends are behind a simulated endpoint-independent NAT, which
+          // requires at least one side to have a srflx candidate to work.
+          await connect(offerer, answerer, 2000, `no ICE servers`);
+          ok(false, "ICE should either have failed, or timed out!");
+        } catch (e) {
+          if (!(e instanceof Error)) throw e;
+          ok(true, "ICE should either have failed, or timed out!");
+        }
+
+        // This _won't_ help, because the answerer does not get to decide to
+        // trigger an ICE restart.
+        answerer.setConfiguration({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+        try {
+          await connectNoTrickleWait(offerer, answerer, 2000, `no ICE servers`);
+          ok(false, "ICE should either have failed, or timed out!");
+        } catch (e) {
+          if (!(e instanceof Error)) throw e;
+          ok(true, "ICE should either have failed, or timed out!");
+        }
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addOffererTurnServerAllowsIceToConnect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.filtering_type', 'PORT_DEPENDENT'],
+          ['media.peerconnection.nat_simulator.mapping_type', 'PORT_DEPENDENT']);
+
+      const offerer = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+      const answerer = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+
+      try {
+        try {
+          // Both ends are behind a simulated port-dependent NAT, which
+          // requires at least one side to have a relay candidate to work.
+          await connect(offerer, answerer, 2000, `just a STUN server`);
+          ok(false, "ICE should either have failed, or timed out!");
+        } catch (e) {
+          if (!(e instanceof Error)) throw e;
+          ok(true, "ICE should either have failed, or timed out!");
+        }
+
+        offerer.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+        await connect(offerer, answerer, 32000, `a TURN (udp) server`);
+      } finally {
+        offerer.close();
+        answerer.close();
+        await SpecialPowers.popPrefEnv();
+      }
+    },
+
+  ];
+
+  runNetworkTest(async () => {
+    const turnServer = iceServersArray.find(server => "username" in server);
+    username = turnServer.username;
+    credential = turnServer.credential;
+    // Just use the first url. It might make sense to look for TURNS first,
+    // since that will always use a hostname, but on CI we don't have TURNS
+    // support anyway (see bug 1323439).
+    const turnHostname = getTurnHostname(turnServer.urls[0]);
+    turnAddressV4 = await dnsLookupV4(turnHostname);
+
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.ice.loopback', true],
+        ['media.getusermedia.insecure.enabled', true]);
+
+    for (const test of tests) {
+      info(`Running test: ${test.name}`);
+      try {
+        await test();
+      } catch (e) {
+        ok(false, `Caught ${e.name}: ${e.message} ${e.stack}`);
+      }
+      info(`Done running test: ${test.name}`);
+      // Make sure we don't build up a pile of GC work, and also get PCImpl to
+      // print their timecards.
+      await new Promise(r => SpecialPowers.exactGC(r));
+    }
+
+    await SpecialPowers.popPrefEnv();
+  }, { useIceServer: true });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithStun300.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithStun300.html
new file mode 100644
index 0000000000..50bc4a6553
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithStun300.html
@@ -0,0 +1,269 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="iceTestUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "857668",
+    title: "RTCPeerConnection check STUN gathering with STUN/300 responses"
+  });
+
+  /* This is pretty hairy, so some background:
+   * Spec is here: https://datatracker.ietf.org/doc/html/rfc8489#section-10
+   * STUN/300 responses allow a server to redirect STUN requests to one or
+     more other servers, as ALTERNATE-SERVER attributes.
+   * The server specifies the IP address, IP version, and port for each
+     ALTERNATE-SERVER.
+   * The spec allows multiple rounds of redirects, and requires the client to
+     remember the servers it has already tried to avoid redirect loops.
+   * For TURNS, the TURN server can also supply an ALTERNATE-DOMAIN attribute,
+     which the client MUST use for the TLS handshake on the new target. The
+     client does _not_ use this as an FQDN; it always uses the address in the
+     ALTERNATE-SERVER. ALTERNATE-DOMAIN is meaningless in the non-TLS case.
+   * STUN/300 with ALTERNATE-SERVER is only defined for the TURN Allocate
+     message type (at least in the context of ICE). Clients are supposed to
+     treat STUN/300 as an unrecoverable error in all other cases. The TURN spec
+     does _not_ spell out how a client should handle multiple ALTERNATE-SERVERs.
+     We just take the first one that we have not already tried, and that is the
+     same IP version that we started with. This is because switching the IP
+     version is problematic for ICE.
+   * The test TURN server opens extra ports that will respond with redirects to
+     the _real_ ports, but the address remains the same. This is because we
+     cannot know ahead of time whether the machine we're running on has more
+     than one IP address of each version. This means the test TURN server is not
+     useful for testing cases where the address changes. Also, the test TURN
+     server does not currently know how to respond with multiple
+     ALTERNATE-SERVERs.
+   * To test cases where the _address_ changes, we instead use a feature in the
+     NAT simulator to respond with fake redirects when the destination address
+     matches an address that we configure with a pref. This feature can add
+     multiple ALTERNATE-SERVERs.
+   * The test TURN server's STUN/300 responses have a proper MESSAGE-INTEGRITY,
+     but the NAT simulator's do _not_. For now, we want both cases to work,
+     because some servers respond with STUN/300 without including
+     MESSAGE-INTEGRITY. This is a spec violation, even though the spec
+     contradicts itself in non-normative language elsewhere.
+   * Right now, neither the NAT simulator nor the test TURN server support
+     ALTERNATE-DOMAIN.
+  */
+
+  // These are the ports the test TURN server will respond with redirects on.
+  // The test TURN server tells us what these are in the JSON it spits out when
+  // we start it.
+  let turnRedirectPort;
+  let turnsRedirectPort;
+
+  // These are the addresses that we will configure the NAT simulator to
+  // redirect to. We do DNS lookups of the host in iceServersArray (provided
+  // by the test TURN server), and put the results here. On some platforms this
+  // will be 127.0.0.1 and ::1, but on others we may use a real address.
+  let redirectTargetV4;
+
+  // Test TURN server tells us these in the JSON it spits out when we start it
+  let username;
+  let credential;
+
+  // This is the address we will configure the NAT simulator to respond with
+  // redirects for. We use an address from TEST-NET since it is really unlikely
+  // we'll see that on a real machine, and also because we do not have
+  // special-case code in nICEr for TEST-NET (like we do for link-local, for
+  // example).
+  const redirectAddressV4 = '198.51.100.1';
+
+  const tests = [
+    async function baselineV4Cases() {
+      await checkSrflx([{urls:[`stun:${redirectTargetV4}`]}]);
+      await checkRelayUdp([{urls:[`turn:${redirectTargetV4}`], username, credential}]);
+      await checkRelayTcp([{urls:[`turn:${redirectTargetV4}?transport=tcp`], username, credential}]);
+      await checkRelayUdpTcp([{urls:[`turn:${redirectTargetV4}`, `turn:${redirectTargetV4}?transport=tcp`], username, credential}]);
+    },
+
+    async function stunV4Redirect() {
+      // This test uses the test TURN server, because nICEr drops responses
+      // without MESSAGE-INTEGRITY on the floor _unless_ they are a STUN/300 to
+      // an Allocate request. If we tried to use the NAT simulator for this, we
+      // would have to wait for nICEr to time out, since the NAT simulator does
+      // not know how to do MESSAGE-INTEGRITY.
+      await checkNoSrflx(
+          [{urls:[`stun:${redirectTargetV4}:${turnRedirectPort}`]}]);
+    },
+
+    async function turnV4UdpPortRedirect() {
+      await checkRelayUdp([{urls:[`turn:${redirectTargetV4}:${turnRedirectPort}`], username, credential}]);
+    },
+
+    async function turnV4TcpPortRedirect() {
+      await checkRelayTcp([{urls:[`turn:${redirectTargetV4}:${turnRedirectPort}?transport=tcp`], username, credential}]);
+    },
+
+    async function turnV4UdpTcpPortRedirect() {
+      await checkRelayUdpTcp([{urls:[`turn:${redirectTargetV4}:${turnRedirectPort}`, `turn:${redirectTargetV4}:${turnRedirectPort}?transport=tcp`], username, credential}]);
+    },
+
+    async function turnV4UdpAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV4}`]);
+      await checkRelayUdp([{urls:[`turn:${redirectAddressV4}`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4TcpAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV4}`]);
+      await checkRelayTcp([{urls:[`turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpTcpAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV4}`]);
+      await checkRelayUdpTcp([{urls:[`turn:${redirectAddressV4}`, `turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpEmptyRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', '']);
+      await checkNoRelay([{urls:[`turn:${redirectAddressV4}`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4TcpEmptyRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', '']);
+      await checkNoRelay([{urls:[`turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpTcpEmptyRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', '']);
+      await checkNoRelay([{urls:[`turn:${redirectAddressV4}`, `turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpAddressAndPortRedirect() {
+      // This should result in two rounds of redirection; the first is by
+      // address, the second is by port.
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV4}`]);
+      await checkRelayUdp([{urls:[`turn:${redirectAddressV4}:${turnRedirectPort}`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4TcpAddressAndPortRedirect() {
+      // This should result in two rounds of redirection; the first is by
+      // address, the second is by port.
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV4}`]);
+      await checkRelayTcp([{urls:[`turn:${redirectAddressV4}:${turnRedirectPort}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpTcpAddressAndPortRedirect() {
+      // This should result in two rounds of redirection; the first is by
+      // address, the second is by port.
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV4}`]);
+      await checkRelayUdpTcp([{urls:[`turn:${redirectAddressV4}:${turnRedirectPort}`, `turn:${redirectAddressV4}:${turnRedirectPort}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpRedirectLoop() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV4}`]);
+      // If we don't detect the loop, gathering will not finish
+      await checkNoRelay([{urls:[`turn:${redirectAddressV4}`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4TcpRedirectLoop() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV4}`]);
+      // If we don't detect the loop, gathering will not finish
+      await checkNoRelay([{urls:[`turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpTcpRedirectLoop() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV4}`]);
+      // If we don't detect the loop, gathering will not finish
+      await checkNoRelay([{urls:[`turn:${redirectAddressV4}`, `turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpMultipleAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV4},${redirectTargetV4}`]);
+      await checkRelayUdp([{urls:[`turn:${redirectAddressV4}`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4TcpMultipleAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV4},${redirectTargetV4}`]);
+      await checkRelayTcp([{urls:[`turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpTcpMultipleAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV4},${redirectTargetV4}`]);
+      await checkRelayUdpTcp([{urls:[`turn:${redirectAddressV4}`, `turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+  ];
+
+  runNetworkTest(async () => {
+    const turnServer = iceServersArray.find(server => "username" in server);
+    username = turnServer.username;
+    credential = turnServer.credential;
+    // Special props, non-standard
+    turnRedirectPort = turnServer.turn_redirect_port;
+    turnsRedirectPort = turnServer.turns_redirect_port;
+    // Just use the first url. It might make sense to look for TURNS first,
+    // since that will always use a hostname, but on CI we don't have TURNS
+    // support anyway (see bug 1323439).
+    const turnHostname = getTurnHostname(turnServer.urls[0]);
+    redirectTargetV4 = await dnsLookupV4(turnHostname);
+
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.ice.loopback', true],
+        ['media.getusermedia.insecure.enabled', true]);
+
+    for (const test of tests) {
+      info(`Running test: ${test.name}`);
+      await test();
+      info(`Done running test: ${test.name}`);
+    }
+
+    await SpecialPowers.popPrefEnv();
+  }, { useIceServer: true });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithStun300IPv6.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithStun300IPv6.html
new file mode 100644
index 0000000000..16f8f39978
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithStun300IPv6.html
@@ -0,0 +1,283 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="iceTestUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "857668",
+    title: "RTCPeerConnection check STUN gathering with STUN/300 responses (IPv6)"
+  });
+
+  /* This is pretty hairy, so some background:
+   * Spec is here: https://datatracker.ietf.org/doc/html/rfc8489#section-10
+   * STUN/300 responses allow a server to redirect STUN requests to one or
+     more other servers, as ALTERNATE-SERVER attributes.
+   * The server specifies the IP address, IP version, and port for each
+     ALTERNATE-SERVER.
+   * The spec allows multiple rounds of redirects, and requires the client to
+     remember the servers it has already tried to avoid redirect loops.
+   * For TURNS, the TURN server can also supply an ALTERNATE-DOMAIN attribute,
+     which the client MUST use for the TLS handshake on the new target. The
+     client does _not_ use this as an FQDN; it always uses the address in the
+     ALTERNATE-SERVER. ALTERNATE-DOMAIN is meaningless in the non-TLS case.
+   * STUN/300 with ALTERNATE-SERVER is only defined for the TURN Allocate
+     message type (at least in the context of ICE). Clients are supposed to
+     treat STUN/300 as an unrecoverable error in all other cases. The TURN spec
+     does _not_ spell out how a client should handle multiple ALTERNATE-SERVERs.
+     We just take the first one that we have not already tried, and that is the
+     same IP version that we started with. This is because switching the IP
+     version is problematic for ICE.
+   * The test TURN server opens extra ports that will respond with redirects to
+     the _real_ ports, but the address remains the same. This is because we
+     cannot know ahead of time whether the machine we're running on has more
+     than one IP address of each version. This means the test TURN server is not
+     useful for testing cases where the address changes. Also, the test TURN
+     server does not currently know how to respond with multiple
+     ALTERNATE-SERVERs.
+   * To test cases where the _address_ changes, we instead use a feature in the
+     NAT simulator to respond with fake redirects when the destination address
+     matches an address that we configure with a pref. This feature can add
+     multiple ALTERNATE-SERVERs.
+   * The test TURN server's STUN/300 responses have a proper MESSAGE-INTEGRITY,
+     but the NAT simulator's do _not_. For now, we want both cases to work,
+     because some servers respond with STUN/300 without including
+     MESSAGE-INTEGRITY. This is a spec violation, even though the spec
+     contradicts itself in non-normative language elsewhere.
+   * Right now, neither the NAT simulator nor the test TURN server support
+     ALTERNATE-DOMAIN.
+  */
+
+  // These are the ports the test TURN server will respond with redirects on.
+  // The test TURN server tells us what these are in the JSON it spits out when
+  // we start it.
+  let turnRedirectPort;
+  let turnsRedirectPort;
+
+  // These are the addresses that we will configure the NAT simulator to
+  // redirect to. We do DNS lookups of the host in iceServersArray (provided
+  // by the test TURN server), and put the results here. On some platforms this
+  // will be 127.0.0.1 and ::1, but on others we may use a real address.
+  let redirectTargetV6;
+
+  // Test TURN server tells us these in the JSON it spits out when we start it
+  let username;
+  let credential;
+
+  // This is the address we will configure the NAT simulator to respond with
+  // redirects for. We use an address from TEST-NET since it is really unlikely
+  // we'll see that on a real machine, and also because we do not have
+  // special-case code in nICEr for TEST-NET (like we do for link-local, for
+  // example).
+  const redirectAddressV6 = '::ffff:198.51.100.1';
+
+  const tests = [
+    async function baselineV6Cases() {
+      await checkSrflx([{urls:[`stun:[${redirectTargetV6}]`]}]);
+      await checkRelayUdp([{urls:[`turn:[${redirectTargetV6}]`], username, credential}]);
+      await checkRelayTcp([{urls:[`turn:[${redirectTargetV6}]?transport=tcp`], username, credential}]);
+      await checkRelayUdpTcp([{urls:[`turn:[${redirectTargetV6}]`, `turn:[${redirectTargetV6}]?transport=tcp`], username, credential}]);
+    },
+
+    async function stunV6Redirect() {
+      // This test uses the test TURN server, because nICEr drops responses
+      // without MESSAGE-INTEGRITY on the floor _unless_ they are a STUN/300 to
+      // an Allocate request. If we tried to use the NAT simulator for this, we
+      // would have to wait for nICEr to time out, since the NAT simulator does
+      // not know how to do MESSAGE-INTEGRITY.
+      await checkNoSrflx(
+          [{urls:[`stun:[${redirectTargetV6}]:${turnRedirectPort}`]}]);
+    },
+
+    async function turnV6UdpPortRedirect() {
+      await checkRelayUdp([{urls:[`turn:[${redirectTargetV6}]:${turnRedirectPort}`], username, credential}]);
+    },
+
+    async function turnV6TcpPortRedirect() {
+      await checkRelayTcp([{urls:[`turn:[${redirectTargetV6}]:${turnRedirectPort}?transport=tcp`], username, credential}]);
+    },
+
+    async function turnV6UdpTcpPortRedirect() {
+      await checkRelayUdpTcp([{urls:[`turn:[${redirectTargetV6}]:${turnRedirectPort}`, `turn:[${redirectTargetV6}]:${turnRedirectPort}?transport=tcp`], username, credential}]);
+    },
+
+    async function turnV6UdpAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV6}`]);
+      await checkRelayUdp([{urls:[`turn:[${redirectAddressV6}]`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6TcpAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV6}`]);
+      await checkRelayTcp([{urls:[`turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpTcpAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV6}`]);
+      await checkRelayUdpTcp([{urls:[`turn:[${redirectAddressV6}]`, `turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpEmptyRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', '']);
+      await checkNoRelay([{urls:[`turn:[${redirectAddressV6}]`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6TcpEmptyRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', '']);
+      await checkNoRelay([{urls:[`turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpTcpEmptyRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', '']);
+      await checkNoRelay([{urls:[`turn:[${redirectAddressV6}]`, `turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpAddressAndPortRedirect() {
+      // This should result in two rounds of redirection; the first is by
+      // address, the second is by port.
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV6}`]);
+      await checkRelayUdp([{urls:[`turn:[${redirectAddressV6}]:${turnRedirectPort}`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6TcpAddressAndPortRedirect() {
+      // This should result in two rounds of redirection; the first is by
+      // address, the second is by port.
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV6}`]);
+      await checkRelayTcp([{urls:[`turn:[${redirectAddressV6}]:${turnRedirectPort}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpTcpAddressAndPortRedirect() {
+      // This should result in two rounds of redirection; the first is by
+      // address, the second is by port.
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV6}`]);
+      await checkRelayUdpTcp([{urls:[`turn:[${redirectAddressV6}]:${turnRedirectPort}`, `turn:[${redirectAddressV6}]:${turnRedirectPort}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpRedirectLoop() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV6}`]);
+      // If we don't detect the loop, gathering will not finish
+      await checkNoRelay([{urls:[`turn:[${redirectAddressV6}]`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6TcpRedirectLoop() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV6}`]);
+      // If we don't detect the loop, gathering will not finish
+      await checkNoRelay([{urls:[`turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpTcpRedirectLoop() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV6}`]);
+      // If we don't detect the loop, gathering will not finish
+      await checkNoRelay([{urls:[`turn:[${redirectAddressV6}]`, `turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpMultipleAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV6},${redirectTargetV6}`]);
+      await checkRelayUdp([{urls:[`turn:[${redirectAddressV6}]`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6TcpMultipleAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV6},${redirectTargetV6}`]);
+      await checkRelayTcp([{urls:[`turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpTcpMultipleAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV6},${redirectTargetV6}`]);
+      await checkRelayUdpTcp([{urls:[`turn:[${redirectAddressV6}]`, `turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+  ];
+
+  runNetworkTest(async () => {
+    const turnServer = iceServersArray.find(server => "username" in server);
+    username = turnServer.username;
+    credential = turnServer.credential;
+    // Special props, non-standard
+    turnRedirectPort = turnServer.turn_redirect_port;
+    turnsRedirectPort = turnServer.turns_redirect_port;
+    // Just use the first url. It might make sense to look for TURNS first,
+    // since that will always use a hostname, but on CI we don't have TURNS
+    // support anyway (see bug 1323439).
+    const turnHostname = getTurnHostname(turnServer.urls[0]);
+
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.ice.loopback', true],
+        ['media.getusermedia.insecure.enabled', true]);
+
+    if (await ipv6Supported()) {
+      redirectTargetV6 = await dnsLookupV6(turnHostname);
+      if (redirectTargetV6 == '' && turnHostname == 'localhost') {
+        // Our testers don't seem to have IPv6 DNS resolution for localhost
+        // set up...
+        redirectTargetV6 = '::1';
+      }
+
+      if (redirectTargetV6 != '') {
+        for (const test of tests) {
+          info(`Running test: ${test.name}`);
+          await test();
+          info(`Done running test: ${test.name}`);
+        }
+      } else {
+        ok(false, `This machine has an IPv6 address, but ${turnHostname} does not resolve to an IPv6 address`);
+      }
+    } else {
+      ok(false, 'This machine appears to not have an IPv6 address');
+    }
+
+    await SpecialPowers.popPrefEnv();
+  }, { useIceServer: true });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_glean.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_glean.html
new file mode 100644
index 0000000000..d5d6026ee5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_glean.html
@@ -0,0 +1,488 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1401592",
+  title: "Test that glean is recording stats as expected",
+  visible: true
+});
+
+const { AppConstants } = SpecialPowers.ChromeUtils.import(
+  "resource://gre/modules/AppConstants.jsm"
+);
+
+async function getAllWarningRates() {
+  return {
+    warnNoGetparameters:
+    await GleanTest.rtcrtpsenderSetparameters.warnNoGetparameters.testGetValue(),
+    warnLengthChanged:
+    await GleanTest.rtcrtpsenderSetparameters.warnLengthChanged.testGetValue(),
+    warnRidChanged:
+    await GleanTest.rtcrtpsenderSetparameters.warnRidChanged.testGetValue(),
+    warnNoTransactionid:
+    await GleanTest.rtcrtpsenderSetparameters.warnNoTransactionid.testGetValue(),
+    warnStaleTransactionid:
+    await GleanTest.rtcrtpsenderSetparameters.warnStaleTransactionid.testGetValue(),
+  };
+}
+
+const tests = [
+  async function checkRTCRtpSenderCount() {
+    const pc = new RTCPeerConnection();
+    const oldCount = await GleanTest.rtcrtpsender.count.testGetValue() ?? 0;
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    const countDiff = await GleanTest.rtcrtpsender.count.testGetValue() - oldCount;
+    is(countDiff, 1, "Glean should have recorded the creation of a single RTCRtpSender");
+  },
+
+  async function checkRTCRtpSenderSetParametersCompatCount() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const oldCount = await GleanTest.rtcrtpsender.countSetparametersCompat.testGetValue() ?? 0;
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    const countDiff = await GleanTest.rtcrtpsender.countSetparametersCompat.testGetValue() - oldCount;
+    is(countDiff, 1, "Glean should have recorded the creation of a single RTCRtpSender that uses the setParameters compat mode");
+  },
+
+  async function checkSendEncodings() {
+    const pc = new RTCPeerConnection();
+    const oldRate = await GleanTest.rtcrtpsender.usedSendencodings.testGetValue();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    const newRate = await GleanTest.rtcrtpsender.usedSendencodings.testGetValue();
+    is(newRate.denominator, oldRate.denominator + 1, "Glean should have recorded the creation of a single RTCRtpSender");
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded the use of sendEncodings");
+  },
+
+  async function checkAddTransceiverNoSendEncodings() {
+    const pc = new RTCPeerConnection();
+    const oldRate = await GleanTest.rtcrtpsender.usedSendencodings.testGetValue();
+    const {sender} = pc.addTransceiver('video');
+    const newRate = await GleanTest.rtcrtpsender.usedSendencodings.testGetValue();
+    is(newRate.denominator, oldRate.denominator + 1, "Glean should have recorded the creation of a single RTCRtpSender");
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded a use of sendEncodings");
+  },
+
+  async function checkAddTrack() {
+    const pc = new RTCPeerConnection();
+    const oldRate = await GleanTest.rtcrtpsender.usedSendencodings.testGetValue();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc.addTrack(stream.getTracks()[0]);
+    const newRate = await GleanTest.rtcrtpsender.usedSendencodings.testGetValue();
+    is(newRate.denominator, oldRate.denominator + 1, "Glean should have recorded the creation of a single RTCRtpSender");
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded a use of sendEncodings");
+  },
+
+  async function checkGoodSetParametersCompatMode() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    const oldWarningRates = await getAllWarningRates();
+    await sender.setParameters(sender.getParameters());
+    const newWarningRates = await getAllWarningRates();
+    isDeeply(oldWarningRates, newWarningRates);
+  },
+
+  async function checkBadSetParametersNoGetParametersWarning() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.warnNoGetparameters.testGetValue();
+    let oldBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameNoGetparameters["example.com"].testGetValue() || 0;
+
+    await sender.setParameters({encodings: [{rid: "0"},{rid: "1"},{rid: "2"}]});
+
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.warnNoGetparameters.testGetValue();
+    let newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameNoGetparameters["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded a warning in setParameters due to lack of a getParameters call");
+
+    if (AppConstants.EARLY_BETA_OR_EARLIER) {
+      is(newBlameCount, oldBlameCount + 1, "Glean should have recorded that example.com encountered this warning");
+    } else {
+      is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning, because we're running this test on a stable channel");
+    }
+
+    oldRate = newRate;
+    oldBlameCount = newBlameCount;
+
+    // Glean should only record the warning once per sender!
+    await sender.setParameters({encodings: [{rid: "0"},{rid: "1"},{rid: "2"}]});
+
+    newRate = await GleanTest.rtcrtpsenderSetparameters.warnNoGetparameters.testGetValue();
+    newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameNoGetparameters["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another warning in setParameters due to lack of a getParameters call");
+    is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning a second time, since this is the same sender");
+  },
+
+  async function checkBadSetParametersLengthChangedWarning() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.warnLengthChanged.testGetValue();
+    let oldBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameLengthChanged["example.com"].testGetValue() || 0;
+
+    let params = sender.getParameters();
+    params.encodings.pop();
+    await sender.setParameters(params);
+
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.warnLengthChanged.testGetValue();
+    let newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameLengthChanged["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded a warning due to a length change in encodings");
+
+    if (AppConstants.EARLY_BETA_OR_EARLIER) {
+      is(newBlameCount, oldBlameCount + 1, "Glean should have recorded that example.com encountered this warning");
+    } else {
+      is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning, because we're running this test on a stable channel");
+    }
+
+    oldRate = newRate;
+    oldBlameCount = newBlameCount;
+
+    // Glean should only record the warning once per sender!
+    params = sender.getParameters();
+    params.encodings.pop();
+    await sender.setParameters(params);
+
+    newRate = await GleanTest.rtcrtpsenderSetparameters.warnLengthChanged.testGetValue();
+    newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameLengthChanged["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another warning due to a length change in encodings");
+    is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning a second time, since this is the same sender");
+  },
+
+  async function checkBadSetParametersRidChangedWarning() {
+    // This pref does not let rid change errors slide anymore
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failRidChanged.testGetValue();
+    let oldWarnRate = await GleanTest.rtcrtpsenderSetparameters.warnRidChanged.testGetValue();
+
+    let params = sender.getParameters();
+    params.encodings[1].rid = "foo";
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failRidChanged.testGetValue();
+    let newWarnRate = await GleanTest.rtcrtpsenderSetparameters.warnRidChanged.testGetValue();
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded an error due to a rid change in encodings");
+    is(newWarnRate.numerator, oldWarnRate.numerator, "Glean should not have recorded a warning due to a rid change in encodings");
+
+    // Glean should only record the error once per sender!
+    params = sender.getParameters();
+    params.encodings[1].rid = "bar";
+    oldRate = newRate;
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failRidChanged.testGetValue();
+    newWarnRate = await GleanTest.rtcrtpsenderSetparameters.warnRidChanged.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another error due to a rid change in encodings");
+    is(newWarnRate.numerator, oldWarnRate.numerator, "Glean should not have recorded a warning due to a rid change in encodings");
+   },
+
+  async function checkBadSetParametersNoTransactionIdWarning() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.warnNoTransactionid.testGetValue();
+    let oldBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameNoTransactionid["example.com"].testGetValue() || 0;
+
+    await sender.setParameters({encodings: [{rid: "0"},{rid: "1"},{rid: "2"}]});
+
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.warnNoTransactionid.testGetValue();
+    let newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameNoTransactionid["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded a warning due to missing transactionId in setParameters");
+
+    if (AppConstants.EARLY_BETA_OR_EARLIER) {
+      is(newBlameCount, oldBlameCount + 1, "Glean should have recorded that example.com encountered this warning");
+    } else {
+      is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning, because we're running this test on a stable channel");
+    }
+
+    oldRate = newRate;
+    oldBlameCount = newBlameCount;
+
+    // Glean should only record the warning once per sender!
+    await sender.setParameters({encodings: [{rid: "0"},{rid: "1"},{rid: "2"}]});
+
+    newRate = await GleanTest.rtcrtpsenderSetparameters.warnNoTransactionid.testGetValue();
+    newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameNoTransactionid["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another warning due to missing transactionId in setParameters");
+    is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning a second time, since this is the same sender");
+  },
+
+  async function checkBadSetParametersStaleTransactionIdWarning() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.warnStaleTransactionid.testGetValue();
+    let oldBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameStaleTransactionid["example.com"].testGetValue() || 0;
+
+    let params = sender.getParameters();
+    // Cause transactionId to be stale
+    await pc.createOffer();
+    // ...but make sure there is a recent getParameters call
+    sender.getParameters();
+    await sender.setParameters(params);
+
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.warnStaleTransactionid.testGetValue();
+    let newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameStaleTransactionid["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded a warning due to stale transactionId in setParameters");
+
+    if (AppConstants.EARLY_BETA_OR_EARLIER) {
+      is(newBlameCount, oldBlameCount + 1, "Glean should have recorded that example.com encountered this warning");
+    } else {
+      is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning, because we're running this test on a stable channel");
+    }
+
+    oldRate = newRate;
+    oldBlameCount = newBlameCount;
+
+    // Glean should only record the warning once per sender!
+    params = sender.getParameters();
+    // Cause transactionId to be stale
+    await pc.createOffer();
+    // ...but make sure there is a recent getParameters call
+    sender.getParameters();
+    await sender.setParameters(params);
+
+    newRate = await GleanTest.rtcrtpsenderSetparameters.warnStaleTransactionid.testGetValue();
+    newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameStaleTransactionid["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another warning due to stale transactionId in setParameters");
+    is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning a second time, since this is the same sender");
+  },
+
+  async function checkBadSetParametersLengthChangedError() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", false]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failLengthChanged.testGetValue();
+    let params = sender.getParameters();
+    params.encodings.pop();
+    try {
+      await sender.setParameters(params);
+    } catch(e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failLengthChanged.testGetValue();
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded an error due to a length change in encodings");
+
+    // Glean should only record the error once per sender!
+    params = sender.getParameters();
+    params.encodings.pop();
+    oldRate = newRate;
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failLengthChanged.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another error due to a length change in encodings");
+  },
+
+  async function checkBadSetParametersRidChangedError() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", false]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failRidChanged.testGetValue();
+    let params = sender.getParameters();
+    params.encodings[1].rid = "foo";
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failRidChanged.testGetValue();
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded an error due to a rid change in encodings");
+
+    // Glean should only record the error once per sender!
+    params = sender.getParameters();
+    params.encodings[1].rid = "bar";
+    oldRate = newRate;
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failRidChanged.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another error due to a rid change in encodings");
+  },
+
+  async function checkBadSetParametersNoGetParametersError() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", false]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failNoGetparameters.testGetValue();
+    try {
+      await sender.setParameters({encodings: [{rid: "0"},{rid: "1"},{rid: "2"}]});
+    } catch (e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failNoGetparameters.testGetValue();
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded an error in setParameters due to lack of a getParameters call");
+
+    // Glean should only record the error once per sender!
+    oldRate = newRate;
+    try {
+      await sender.setParameters({encodings: [{rid: "0"},{rid: "1"},{rid: "2"}]});
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failNoGetparameters.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another error in setParameters due to lack of a getParameters call");
+  },
+
+  async function checkBadSetParametersStaleTransactionIdError() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", false]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failStaleTransactionid.testGetValue();
+    let params = sender.getParameters();
+    // Cause transactionId to be stale
+    await pc.createOffer();
+    // ...but make sure there is a recent getParameters call
+    sender.getParameters();
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failStaleTransactionid.testGetValue();
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded an error due to stale transactionId in setParameters");
+
+    // Glean should only record the error once per sender!
+    oldRate = newRate;
+    params = sender.getParameters();
+    // Cause transactionId to be stale
+    await pc.createOffer();
+    // ...but make sure there is a recent getParameters call
+    sender.getParameters();
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failStaleTransactionid.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another error due to stale transactionId in setParameters");
+  },
+
+  async function checkBadSetParametersNoEncodingsError() {
+    // If we do not allow the old setParameters, this will fail the length check
+    // instead.
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failNoEncodings.testGetValue();
+    let params = sender.getParameters();
+    params.encodings = [];
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failNoEncodings.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded an error due to empty encodings in setParameters");
+
+    // Glean should only record the error once per sender!
+    oldRate = newRate;
+    params = sender.getParameters();
+    params.encodings = [];
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failNoEncodings.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded an error due empty encodings in setParameters");
+  },
+
+  async function checkBadSetParametersOtherError() {
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failOther.testGetValue();
+    let params = sender.getParameters();
+    params.encodings[0].scaleResolutionDownBy = 0.5;
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failOther.testGetValue();
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded an error due to some other failure");
+
+    // Glean should only record the error once per sender!
+    oldRate = newRate;
+    params = sender.getParameters();
+    params.encodings[0].scaleResolutionDownBy = 0.5;
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failOther.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another error due to some other failure");
+  },
+
+];
+
+runNetworkTest(async () => {
+  for (const test of tests) {
+    info(`Running test: ${test.name}`);
+    await test();
+    info(`Done running test: ${test.name}`);
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_iceFailure.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_iceFailure.html
new file mode 100644
index 0000000000..1b82473997
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_iceFailure.html
@@ -0,0 +1,84 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1087629",
+    title: "Wait for ICE failure"
+  });
+
+// Test iceFailure
+
+function PC_LOCAL_SETUP_NULL_ICE_HANDLER(test) {
+  test.pcLocal.setupIceCandidateHandler(test, function() {}, function () {});
+}
+function PC_REMOTE_SETUP_NULL_ICE_HANDLER(test) {
+  test.pcRemote.setupIceCandidateHandler(test, function() {}, function () {});
+}
+function PC_REMOTE_ADD_FAKE_ICE_CANDIDATE(test) {
+  var cand = {"candidate":"candidate:0 1 UDP 2130379007 192.0.2.1 12345 typ host","sdpMid":"","sdpMLineIndex":0};
+  test.pcRemote.storeOrAddIceCandidate(cand);
+  info(test.pcRemote + " Stored fake candidate: " + JSON.stringify(cand));
+}
+function PC_LOCAL_ADD_FAKE_ICE_CANDIDATE(test) {
+  var cand = {"candidate":"candidate:0 1 UDP 2130379007 192.0.2.2 56789 typ host","sdpMid":"","sdpMLineIndex":0};
+    test.pcLocal.storeOrAddIceCandidate(cand);
+  info(test.pcLocal + " Stored fake candidate: " + JSON.stringify(cand));
+}
+function PC_LOCAL_WAIT_FOR_ICE_FAILURE(test) {
+  return test.pcLocal.iceFailed.then(() => {
+    ok(true, this.pcLocal + " Ice Failure Reached.");
+	});
+}
+function PC_REMOTE_WAIT_FOR_ICE_FAILURE(test) {
+  return test.pcRemote.iceFailed.then(() => {
+    ok(true, this.pcRemote + " Ice Failure Reached.");
+    });
+}
+function PC_LOCAL_WAIT_FOR_ICE_FAILED(test) {
+  var resolveIceFailed;
+  test.pcLocal.iceFailed = new Promise(r => resolveIceFailed = r);
+  test.pcLocal.ice_connection_callbacks.checkIceStatus = () => {
+	if (test.pcLocal._pc.iceConnectionState === "failed") {
+	  resolveIceFailed();
+	}
+  }
+}
+function PC_REMOTE_WAIT_FOR_ICE_FAILED(test) {
+  var resolveIceFailed;
+  test.pcRemote.iceFailed = new Promise(r => resolveIceFailed = r);
+  test.pcRemote.ice_connection_callbacks.checkIceStatus = () => {
+	if (test.pcRemote._pc.iceConnectionState === "failed") {
+	  resolveIceFailed();
+	}
+  }
+}
+
+runNetworkTest(async () => {
+  await pushPrefs(
+    ['media.peerconnection.ice.stun_client_maximum_transmits', 3],
+    ['media.peerconnection.ice.trickle_grace_period', 3000],
+  );
+  var test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.replace("PC_LOCAL_SETUP_ICE_HANDLER", PC_LOCAL_SETUP_NULL_ICE_HANDLER);
+  test.chain.replace("PC_REMOTE_SETUP_ICE_HANDLER", PC_REMOTE_SETUP_NULL_ICE_HANDLER);
+  test.chain.insertAfter("PC_REMOTE_SETUP_NULL_ICE_HANDLER", PC_LOCAL_WAIT_FOR_ICE_FAILED);
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_ICE_FAILED", PC_REMOTE_WAIT_FOR_ICE_FAILED);
+  test.chain.removeAfter("PC_LOCAL_SET_REMOTE_DESCRIPTION");
+  test.chain.append([
+    PC_REMOTE_ADD_FAKE_ICE_CANDIDATE,
+    PC_LOCAL_ADD_FAKE_ICE_CANDIDATE,
+    PC_LOCAL_WAIT_FOR_ICE_FAILURE,
+    PC_REMOTE_WAIT_FOR_ICE_FAILURE
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_insertDTMF.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_insertDTMF.html
new file mode 100644
index 0000000000..ca8b866a6d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_insertDTMF.html
@@ -0,0 +1,76 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1291715",
+  title: "Test insertDTMF on sender",
+  visible: true
+});
+
+function insertdtmftest(pc) {
+  ok(pc.getSenders().length, "have senders");
+  var sender = pc.getSenders()[0];
+  ok(sender.dtmf, "sender has dtmf object");
+
+  ok(sender.dtmf.toneBuffer === "", "sender should start with empty tonebuffer");
+
+  // These will trigger assertions on debug builds if we do not enforce the
+  // specified minimums and maximums for duration and interToneGap.
+  sender.dtmf.insertDTMF("A", 10);
+  sender.dtmf.insertDTMF("A", 10000);
+  sender.dtmf.insertDTMF("A", 70, 10);
+
+  var threw = false;
+  try {
+    sender.dtmf.insertDTMF("bad tones");
+  } catch (ex) {
+    threw = true;
+    is(ex.code, DOMException.INVALID_CHARACTER_ERR, "Expected InvalidCharacterError");
+  }
+  ok(threw, "Expected exception");
+
+  sender.dtmf.insertDTMF("A");
+  sender.dtmf.insertDTMF("B");
+  ok(!sender.dtmf.toneBuffer.includes("A"), "calling insertDTMF should replace current characters");
+
+  sender.dtmf.insertDTMF("aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa");
+  ok(sender.dtmf.toneBuffer.includes("A"), "lowercase characters should be normalized");
+
+  pc.removeTrack(sender);
+  threw = false;
+  try {
+    sender.dtmf.insertDTMF("AAA");
+  } catch (ex) {
+    threw = true;
+    is(ex.code, DOMException.INVALID_STATE_ERR, "Expected InvalidStateError");
+  }
+  ok(threw, "Expected exception");
+}
+
+runNetworkTest(() => {
+  test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW");
+
+  // Test sender dtmf.
+  test.chain.append([
+    function PC_LOCAL_INSERT_DTMF(test) {
+      // We want to call removeTrack
+      test.pcLocal.expectNegotiationNeeded();
+      return insertdtmftest(test.pcLocal._pc);
+    }
+  ]);
+
+  return pushPrefs(['media.peerconnection.dtmf.enabled', true])
+    .then(() => test.run());
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_localReofferRollback.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_localReofferRollback.html
new file mode 100644
index 0000000000..16406ece6e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_localReofferRollback.html
@@ -0,0 +1,44 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "952145",
+    title: "Rollback local reoffer"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain, [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}, {audio: true}],
+                                   [{audio: true}]);
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+
+        function PC_LOCAL_CREATE_AND_SET_OFFER(test) {
+          return test.createOffer(test.pcLocal).then(offer => {
+            return test.setLocalDescription(test.pcLocal, offer, HAVE_LOCAL_OFFER);
+          });
+        },
+
+        function PC_LOCAL_ROLLBACK(test) {
+          // the negotiationNeeded slot should have been true both before and
+          // after this SLD, so the event should fire again.
+          test.pcLocal.expectNegotiationNeeded();
+          return test.setLocalDescription(test.pcLocal,
+                                          { type: "rollback", sdp: "" },
+                                          STABLE);
+        },
+    ]);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_localRollback.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_localRollback.html
new file mode 100644
index 0000000000..5bdc8cc029
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_localRollback.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "952145",
+    title: "Rollback local offer"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    test.chain.insertBefore('PC_LOCAL_CREATE_OFFER', [
+        function PC_REMOTE_CREATE_AND_SET_OFFER(test) {
+          return test.createOffer(test.pcRemote).then(offer => {
+            return test.setLocalDescription(test.pcRemote, offer, HAVE_LOCAL_OFFER);
+          });
+        },
+
+        function PC_REMOTE_ROLLBACK(test) {
+          // the negotiationNeeded slot should have been true both before and
+          // after this SLD, so the event should fire again.
+          test.pcRemote.expectNegotiationNeeded();
+          return test.setLocalDescription(test.pcRemote,
+                                          { type: "rollback", sdp: "" },
+                                          STABLE);
+        },
+
+        // Rolling back should shut down gathering
+        function PC_REMOTE_WAIT_FOR_END_OF_TRICKLE(test) {
+          return test.pcRemote.endOfTrickleIce;
+        },
+
+        function PC_REMOTE_SETUP_ICE_HANDLER(test) {
+          test.pcRemote.setupIceCandidateHandler(test);
+        },
+    ]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_maxFsConstraint.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_maxFsConstraint.html
new file mode 100644
index 0000000000..a2f2555020
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_maxFsConstraint.html
@@ -0,0 +1,112 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1393687",
+    title: "Enforce max-fs constraint on a PeerConnection",
+    visible: true
+  });
+
+  var mustRejectWith = (msg, reason, f) =>
+    f().then(() => ok(false, msg),
+             e => is(e.name, reason, msg));
+
+  var removeAllButCodec = (d, codec) =>
+    (d.sdp = d.sdp.replace(/m=video (\w) UDP\/TLS\/RTP\/SAVPF \w.*\r\n/,
+                           "m=video $1 UDP/TLS/RTP/SAVPF " + codec + "\r\n"), d);
+
+  var mungeSDP = (d, forceH264) => {
+    if (forceH264) {
+      removeAllButCodec(d, 126);
+      d.sdp = d.sdp.replace(/a=fmtp:126 (.*);packetization-mode=1/, "a=fmtp:126 $1;packetization-mode=1;max-fs=100");
+    } else {
+      d.sdp = d.sdp.replace(/max-fs=\d+/, "max-fs=100");
+    }
+    return d;
+  };
+
+  const checkForH264Support = async () => {
+    const pc = new RTCPeerConnection();
+    const offer = await pc.createOffer({offerToReceiveVideo: true});
+    return offer.sdp.match(/a=rtpmap:[1-9][0-9]* H264/);
+  };
+
+  let resolutionAlignment = 1;
+
+  function testScale(codec) {
+    var v1 = createMediaElement('video', 'v1');
+    var v2 = createMediaElement('video', 'v2');
+
+    var pc1 = new RTCPeerConnection();
+    var pc2 = new RTCPeerConnection();
+
+    var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
+    pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+    info("testing max-fs with" + codec);
+
+    pc1.onnegotiationneeded = e =>
+      pc1.createOffer()
+      .then(d => pc1.setLocalDescription(mungeSDP(d, codec == "H264")))
+      .then(() => pc2.setRemoteDescription(pc1.localDescription))
+      .then(() => pc2.createAnswer()).then(d => pc2.setLocalDescription(mungeSDP(d, codec =="H264")))
+      .then(() => pc1.setRemoteDescription(pc2.localDescription))
+      .catch(generateErrorCallback());
+
+    pc2.ontrack = e => {
+      v2.srcObject = e.streams[0];
+    };
+
+    var stream;
+
+    return navigator.mediaDevices.getUserMedia({ video: true })
+      .then(s => {
+        stream = s;
+        v1.srcObject = stream;
+        let track = stream.getVideoTracks()[0];
+        let sender = pc1.addTrack(track, stream);
+        is(v2.currentTime, 0, "v2.currentTime is zero at outset");
+      })
+      .then(() => wait(5000))
+      .then(() => {
+        if (v2.videoWidth == 0 && v2.videoHeight == 0) {
+          info("Skipping test, insufficient time for video to start.");
+        } else {
+          const expectedWidth = 184 - 184 % resolutionAlignment;
+          const expectedHeight = 138 - 138 % resolutionAlignment;
+          is(v2.videoWidth, expectedWidth,
+             `sink width should be ${expectedWidth} for ${codec}`);
+          is(v2.videoHeight, expectedHeight,
+             `sink height should be ${expectedHeight} for ${codec}`);
+        }})
+      .then(() => {
+        stream.getTracks().forEach(track => track.stop());
+        v1.srcObject = v2.srcObject = null;
+      }).catch(generateErrorCallback());
+  }
+
+  runNetworkTest(async () => {
+    await pushPrefs(['media.peerconnection.video.lock_scaling', true]);
+    if (await checkForH264Support()) {
+      if (navigator.userAgent.includes("Android")) {
+        // Android only has a hw encoder for h264
+        resolutionAlignment = 16;
+      }
+      await matchPlatformH264CodecPrefs();
+      await testScale("H264");
+    }
+
+    // Disable h264 hardware support, to ensure it is not prioritized over VP8
+    await pushPrefs(["media.webrtc.hw.h264.enabled", false]);
+    await testScale("VP8");
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_multiple_captureStream_canvas_2d.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_multiple_captureStream_canvas_2d.html
new file mode 100644
index 0000000000..9ad25e7852
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_multiple_captureStream_canvas_2d.html
@@ -0,0 +1,115 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1166832",
+  title: "Canvas(2D)::Multiple CaptureStream as video-only input to peerconnection",
+  visible: true
+});
+
+/**
+ * Test to verify using multiple capture streams concurrently.
+ */
+runNetworkTest(async () => {
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false]);
+    await pushPrefs(["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  var test = new PeerConnectionTest();
+  var h = new CaptureStreamTestHelper2D(50, 50);
+
+  var vremote1;
+  var stream1;
+  var canvas1 = h.createAndAppendElement('canvas', 'source_canvas1');
+
+  var vremote2;
+  var stream2;
+  var canvas2 = h.createAndAppendElement('canvas', 'source_canvas2');
+
+  const threshold = 128;
+
+  test.setMediaConstraints([{video: true}, {video: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function PC_LOCAL_CANVAS_CAPTURESTREAM(test) {
+      h.drawColor(canvas1, h.green);
+      h.drawColor(canvas2, h.blue);
+      stream1 = canvas1.captureStream(0); // fps = 0 to capture single frame
+      test.pcLocal.attachLocalStream(stream1);
+      stream2 = canvas2.captureStream(0); // fps = 0 to capture single frame
+      test.pcLocal.attachLocalStream(stream2);
+      var i = 0;
+      return setInterval(function() {
+        try {
+          info("draw " + i ? "green" : "red/blue");
+          h.drawColor(canvas1, i ? h.green : h.red);
+          h.drawColor(canvas2, i ? h.green : h.blue);
+          i = 1 - i;
+          stream1.requestFrame();
+          stream2.requestFrame();
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
+    }
+  ]);
+
+  test.chain.append([
+    function CHECK_REMOTE_VIDEO() {
+      is(test.pcRemote.remoteMediaElements.length, 2, "pcRemote Should have 2 remote media elements");
+      vremote1 = test.pcRemote.remoteMediaElements[0];
+      vremote2 = test.pcRemote.remoteMediaElements[1];
+
+      // since we don't know which remote video is created first, we don't know
+      // which should be blue or red, but this will make sure that one is
+      // green and one is blue
+      return Promise.race([
+               Promise.all([
+                 h.pixelMustBecome(vremote1, h.red, {
+                   threshold,
+                   infoString: "pcRemote's remote1 should become red",
+                 }),
+                 h.pixelMustBecome(vremote2, h.blue, {
+                   threshold,
+                   infoString: "pcRemote's remote2 should become blue",
+                 }),
+               ]),
+               Promise.all([
+                 h.pixelMustBecome(vremote2, h.red, {
+                   threshold,
+                   infoString: "pcRemote's remote2 should become red",
+                 }),
+                 h.pixelMustBecome(vremote1, h.blue, {
+                   threshold,
+                   infoString: "pcRemote's remote1 should become blue",
+                 }),
+               ])
+             ]);
+    },
+    function WAIT_FOR_REMOTE_BOTH_GREEN() {
+      return Promise.all([
+               h.pixelMustBecome(vremote1, h.green, {
+                   threshold,
+                   infoString: "pcRemote's remote1 should become green",
+               }),
+               h.pixelMustBecome(vremote2, h.green, {
+                   threshold,
+                   infoString: "pcRemote's remote2 should become green",
+               }),
+             ])
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleAnswer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleAnswer.html
new file mode 100644
index 0000000000..7e3fd78430
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleAnswer.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1060102",
+    title: "Basic audio only SDP answer without trickle ICE"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    makeAnswererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleOffer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleOffer.html
new file mode 100644
index 0000000000..12b2a95596
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleOffer.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1060102",
+    title: "Basic audio only SDP offer without trickle ICE"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    makeOffererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleOfferAnswer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleOfferAnswer.html
new file mode 100644
index 0000000000..554750e975
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleOfferAnswer.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1060102",
+    title: "Basic audio only SDP offer and answer without trickle ICE"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    makeOffererNonTrickle(test.chain);
+    makeAnswererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_nonDefaultRate.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_nonDefaultRate.html
new file mode 100644
index 0000000000..ad9414cef2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_nonDefaultRate.html
@@ -0,0 +1,200 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({ title: "PeerConnection feed to a graph with non default rate", bug: "1387454" });
+  /**
+   * Run a test to verify that when we use the streem with nonDefault rate to/from a PC
+   * the connection fails. (PC is always on default rate).
+   */
+
+  let pc1;
+  let pc2;
+
+  const offerOptions = {
+    offerToReceiveAudio: 1,
+  };
+
+  function getName(pc) {
+    return (pc === pc1) ? 'pc1' : 'pc2';
+  }
+
+  function getOtherPc(pc) {
+    return (pc === pc1) ? pc2 : pc1;
+  }
+
+  function onAddIceCandidateSuccess(pc) {
+    ok(true, getName(pc) + ' addIceCandidate success');
+  }
+
+  function onAddIceCandidateError(pc, error) {
+    ok(false, getName(pc) + ' failed to add ICE Candidate: ' + error.toString());
+  }
+
+  function onIceCandidate(pc, event, done) {
+    if (!event.candidate) {
+      ok(pc.iceGatheringState === 'complete', getName(pc) + " ICE gathering state has reached complete");
+      done();
+      return;
+    }
+    getOtherPc(pc).addIceCandidate(event.candidate)
+    .then(() => {
+        onAddIceCandidateSuccess(pc);
+      },
+      (err) => {
+        onAddIceCandidateError(pc, err);
+      });
+    info(getName(pc) + ' ICE candidate: ' + event.candidate.candidate);
+  }
+
+  function onIceStateChange(pc, event) {
+    if (pc) {
+      info(getName(pc) + ' ICE state: ' + pc.iceConnectionState);
+      info('ICE state change event: ', event);
+    }
+  }
+
+  function onCreateOfferSuccess(desc) {
+    info('Offer from pc1\n' + desc.sdp);
+    info('pc1 setLocalDescription start');
+
+    pc1.setLocalDescription(desc)
+    .then(() => {
+      onSetLocalSuccess(pc1);
+    },
+    onSetSessionDescriptionError);
+
+    info('pc2 setRemoteDescription start');
+    pc2.setRemoteDescription(desc).then(() => {
+      onSetRemoteSuccess(pc2);
+    },
+    onSetSessionDescriptionError);
+
+    info('pc2 createAnswer start');
+
+    // Since the 'remote' side has no media stream we need
+    // to pass in the right constraints in order for it to
+    // accept the incoming offer of audio and video.
+    pc2.createAnswer()
+    .then(onCreateAnswerSuccess, onCreateSessionDescriptionError);
+  }
+
+  function onSetSessionDescriptionError(error) {
+    ok(false, 'Failed to set session description: ' + error.toString());
+  }
+
+  function onSetLocalSuccess(pc) {
+    ok(true, getName(pc) + ' setLocalDescription complete');
+  }
+
+  function onCreateSessionDescriptionError(error) {
+    ok(false, 'Failed to create session description: ' + error.toString());
+  }
+
+  function onSetRemoteSuccess(pc) {
+    ok(true, getName(pc) + ' setRemoteDescription complete');
+  }
+
+  function onCreateAnswerSuccess(desc) {
+    info('Answer from pc2:\n' + desc.sdp);
+    info('pc2 setLocalDescription start');
+    pc2.setLocalDescription(desc).then(() => {
+      onSetLocalSuccess(pc2);
+    },
+    onSetSessionDescriptionError);
+    info('pc1 setRemoteDescription start');
+    pc1.setRemoteDescription(desc).then(() => {
+        onSetRemoteSuccess(pc1);
+    },
+    onSetSessionDescriptionError);
+  }
+
+  async function getRemoteStream(localStream) {
+    info("got local stream")
+    const audioTracks = localStream.getAudioTracks();
+
+    const servers = null;
+
+    pc1 = new RTCPeerConnection(servers);
+    info('Created local peer connection object pc1');
+    const iceComplete1 = new Promise((resolve, reject) => {
+      pc1.onicecandidate = (e) => {
+        onIceCandidate(pc1, e, resolve);
+      };
+    });
+
+    pc2 = new RTCPeerConnection(servers);
+    info('Created remote peer connection object pc2');
+    const iceComplete2 = new Promise((resolve, reject) => {
+      pc2.onicecandidate = (e) => {
+        onIceCandidate(pc2, e, resolve);
+      };
+    });
+
+    pc1.oniceconnectionstatechange = (e) => {
+      onIceStateChange(pc1, e);
+    };
+    pc2.oniceconnectionstatechange = (e) => {
+      onIceStateChange(pc2, e);
+    };
+
+    const remoteStreamPromise = new Promise((resolve, reject) => {
+      pc2.ontrack = (e) => {
+        info('pc2 received remote stream ' + e.streams[0]);
+        resolve(e.streams[0]);
+      };
+    });
+
+    localStream.getTracks().forEach((track) => {
+      pc1.addTrack(track, localStream);
+    });
+    info('Added local stream to pc1');
+
+    info('pc1 createOffer start');
+    pc1.createOffer(offerOptions)
+      .then(onCreateOfferSuccess,onCreateSessionDescriptionError);
+
+    let promise_arr = await Promise.all([remoteStreamPromise, iceComplete1, iceComplete2]);
+    return promise_arr[0];
+  }
+
+  runTest(async () => {
+    // Local stream operates at non default rate (32000)
+    const nonDefaultRate = 32000;
+    const nonDefault_ctx = new AudioContext({sampleRate: nonDefaultRate});
+    oscillator = nonDefault_ctx.createOscillator();
+    const dest = nonDefault_ctx.createMediaStreamDestination();
+    oscillator.connect(dest);
+    oscillator.start();
+
+    // Wait for remote stream
+    const remoteStream = await getRemoteStream(dest.stream)
+    ok(true, 'Got remote stream ' + remoteStream);
+
+    // remoteStream now comes from PC so operates at default
+    // rates. Verify that by adding to a default context
+    const ac = new AudioContext;
+    const source_default_rate = ac.createMediaStreamSource(remoteStream);
+
+    // Now try to add the remoteStream on a non default context
+    mustThrowWith(
+      "Connect stream with graph of different sample rate",
+      "NotSupportedError", () => {
+        nonDefault_ctx.createMediaStreamSource(remoteStream);
+      }
+    );
+
+    // Close peer connections to make sure we don't get error:
+    // "logged result after SimpleTest.finish(): pc1 addIceCandidate success"
+    // See Bug 1626814.
+    pc1.close();
+    pc2.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveAudio.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveAudio.html
new file mode 100644
index 0000000000..1f936714f1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveAudio.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "850275",
+    title: "Simple offer media constraint test with audio"
+  });
+
+  runNetworkTest(function() {
+    var test = new PeerConnectionTest();
+    test.setMediaConstraints([], [{audio: true}]);
+    test.setOfferOptions({ offerToReceiveAudio: true });
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveVideo.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveVideo.html
new file mode 100644
index 0000000000..c5afbb5c1f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveVideo.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "850275",
+    title: "Simple offer media constraint test with video"
+  });
+
+  runNetworkTest(function() {
+    var test = new PeerConnectionTest();
+    test.setMediaConstraints([], [{video: true}]);
+    test.setOfferOptions({ offerToReceiveVideo: true });
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveVideoAudio.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveVideoAudio.html
new file mode 100644
index 0000000000..d7bc29c6d3
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveVideoAudio.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "850275",
+    title: "Simple offer media constraint test with video/audio"
+  });
+
+  runNetworkTest(function() {
+    var test = new PeerConnectionTest();
+    test.setMediaConstraints([], [{audio: true, video: true}]);
+    test.setOfferOptions({ offerToReceiveVideo: true, offerToReceiveAudio: true });
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_portRestrictions.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_portRestrictions.html
new file mode 100644
index 0000000000..7cd695ff54
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_portRestrictions.html
@@ -0,0 +1,63 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1677046",
+    title: "RTCPeerConnection check restricted ports"
+  });
+
+var makePC = (config, expected_error) => {
+  var exception;
+  try {
+    new RTCPeerConnection(config).close();
+  } catch (e) {
+    exception = e;
+  }
+  is((exception? exception.name : "success"), expected_error || "success",
+     "RTCPeerConnection(" + JSON.stringify(config) + ")");
+};
+
+// This is a test of the iceServers parsing code + readable errors
+runNetworkTest(() => {
+  var exception = null;
+
+  // check various ports on the blocklist
+  makePC({ iceServers: [
+    { urls:"turn:[::1]:6666", username:"p", credential:"p" }] }, "NS_ERROR_UNEXPECTED");
+  makePC({ iceServers: [
+    { urls:"turns:localhost:6667?transport=udp", username:"p", credential:"p" }] },
+    "NS_ERROR_UNEXPECTED");
+  makePC({ iceServers: [
+    { urls:"stun:localhost:21", foo:"" }] }, "NS_ERROR_UNEXPECTED");
+  makePC({ iceServers: [
+    { urls:"stun:[::1]:22", foo:"" }] }, "NS_ERROR_UNEXPECTED");
+  makePC({ iceServers: [
+    { urls:"turn:localhost:5060", username:"p", credential:"p" }] },
+    "NS_ERROR_UNEXPECTED");
+
+  // check various ports on the good list for webrtc (or default port)
+  makePC({ iceServers: [
+    { urls:"turn:[::1]:53", username:"p", credential:"p" },
+    { urls:"turn:[::1]:5349", username:"p", credential:"p" },
+    { urls:"turn:[::1]:3478", username:"p", credential:"p" },
+    { urls:"turn:[::1]", username:"p", credential:"p" },
+    { urls:"turn:localhost:53?transport=udp", username:"p", credential:"p" },
+    { urls:"turn:localhost:3478?transport=udp", username:"p", credential:"p" },
+    { urls:"turn:localhost:53?transport=tcp", username:"p", credential:"p" },
+    { urls:"turn:localhost:3478?transport=tcp", username:"p", credential:"p" },
+    { urls:"turns:localhost:3478?transport=udp", username:"p", credential:"p" },
+    { urls:"stun:localhost", foo:"" }
+  ]});
+
+  // not in the known good ports and not on the generic block list
+  makePC({ iceServers: [{ urls:"turn:localhost:6664", username:"p", credential:"p" }] });
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_promiseSendOnly.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_promiseSendOnly.html
new file mode 100644
index 0000000000..a3fbb5753c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_promiseSendOnly.html
@@ -0,0 +1,61 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1091898",
+    title: "PeerConnection with promises (sendonly)",
+    visible: true
+  });
+
+  var pc1 = new RTCPeerConnection();
+  var pc2 = new RTCPeerConnection();
+
+  var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+  pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
+  pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+  var v1, v2;
+  var delivered = new Promise(resolve => pc2.ontrack = e => {
+    // Test RTCTrackEvent here.
+    ok(e.streams.length, "has streams");
+    ok(e.streams[0].getTrackById(e.track.id), "has track");
+    ok(pc2.getReceivers().some(receiver => receiver == e.receiver), "has receiver");
+    if (e.streams[0].getTracks().length == 2) {
+      // Test RTCTrackEvent required args here.
+      mustThrowWith("RTCTrackEvent wo/required args",
+                    "TypeError", () => new RTCTrackEvent("track", {}));
+      v2.srcObject = e.streams[0];
+      resolve();
+    }
+  });
+
+  runNetworkTest(function() {
+    v1 = createMediaElement('video', 'v1');
+    v2 = createMediaElement('video', 'v2');
+    var canPlayThrough = new Promise(resolve => v2.canplaythrough = e => resolve());
+
+    is(v2.currentTime, 0, "v2.currentTime is zero at outset");
+
+    return navigator.mediaDevices.getUserMedia({ video: true, audio: true })
+    .then(stream => (v1.srcObject = stream).getTracks().forEach(t => pc1.addTrack(t, stream)))
+    .then(() => pc1.createOffer({})) // check that createOffer accepts arg.
+    .then(offer => pc1.setLocalDescription(offer))
+    .then(() => pc2.setRemoteDescription(pc1.localDescription))
+    .then(() => pc2.createAnswer({}))  // check that createAnswer accepts arg.
+    .then(answer => pc2.setLocalDescription(answer))
+    .then(() => pc1.setRemoteDescription(pc2.localDescription))
+    .then(() => delivered)
+//    .then(() => canPlayThrough)    // why doesn't this fire?
+    .then(() => waitUntil(() => v2.currentTime > 0))
+    .then(() => ok(v2.currentTime > 0, "v2.currentTime is moving (" + v2.currentTime + ")"))
+    .then(() => ok(true, "Connected."));
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_recordReceiveTrack.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_recordReceiveTrack.html
new file mode 100644
index 0000000000..d5cb91b048
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_recordReceiveTrack.html
@@ -0,0 +1,101 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+<script src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script>
+createHTML({
+  bug: "1212237",
+  title: "Recording a fresh receive track should not throw",
+  visible: true,
+});
+
+/**
+ * Called when a fresh track is available, and tests that the track can be
+ * recorded until it ends without any thrown errors or fired error events.
+ */
+let generation = 0;
+async function testTrackAccessible(track) {
+  const id = ++generation;
+  info(`Testing accessibility for ${track.kind} track ${id}`);
+  const recorder = new MediaRecorder(new MediaStream([track]));
+  recorder.start();
+  let haveError = new Promise((_, rej) => recorder.onerror = e => rej(e.error));
+  await Promise.race([
+    new Promise(r => recorder.onstart = r),
+    haveError,
+  ]);
+  info(`Recording of ${track.kind} track ${id} started`);
+
+  const {data} = await Promise.race([
+    new Promise(r => recorder.ondataavailable = r),
+    haveError,
+  ]);
+  info(`Recording of ${track.kind} track ${id} finished at size ${data.size}`);
+
+  await Promise.race([
+    new Promise(r => recorder.onstop = r),
+    haveError,
+  ]);
+  info(`Recording of ${track.kind} track ${id} stopped`);
+
+  const element = createMediaElement(track.kind, `recording_${track.id}`);
+  const url = URL.createObjectURL(data);
+  try {
+    element.src = url;
+    element.preload = "metadata";
+    haveError = new Promise(
+      (_, rej) => element.onerror = e => rej(element.error));
+    await Promise.race([
+      new Promise(r => element.onloadeddata = r),
+      haveError,
+    ]);
+    info(`Playback of recording of ${track.kind} track ${id} loaded data`);
+
+    element.play();
+    await Promise.race([
+      new Promise(r => element.onended = r),
+      haveError,
+    ]);
+    info(`Playback of recording of ${track.kind} track ${id} ended`);
+  } finally {
+    URL.revokeObjectURL(data);
+  }
+}
+
+runNetworkTest(async options => {
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+  const test = new PeerConnectionTest(options);
+  test.setMediaConstraints([{video: true}], [{audio: true}]);
+  test.setOfferOptions({offerToReceiveAudio: true});
+  const freshVideoTrackIsAccessible = new Promise(
+      r => test.pcRemote._pc.addEventListener("track", r, {once: true})
+    ).then(({track}) => testTrackAccessible(track));
+  const freshAudioTrackIsAccessible = new Promise(
+      r => test.pcLocal._pc.addEventListener("track", r, {once: true})
+    ).then(({track}) => testTrackAccessible(track));
+  test.chain.append([
+    function PC_CLOSE_TO_END_TRACKS() {
+      return test.close();
+    },
+    async function FRESH_VIDEO_TRACK_IS_ACCESSIBLE() {
+      await freshVideoTrackIsAccessible;
+      ok(true, "A freshly received video track is accessible by MediaRecorder");
+    },
+    async function FRESH_AUDIO_TRACK_IS_ACCESSIBLE() {
+      await freshAudioTrackIsAccessible;
+      ok(true, "A freshly received audio track is accessible by MediaRecorder");
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_relayOnly.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_relayOnly.html
new file mode 100644
index 0000000000..3b07783c04
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_relayOnly.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1187775",
+  title: "peer connection ICE fails on relay-only without TURN"
+});
+
+function PC_LOCAL_NO_CANDIDATES(test) {
+  var isnt = can => is(can, null, "No candidates: " + JSON.stringify(can));
+  test.pcLocal._pc.addEventListener("icecandidate", e => isnt(e.candidate));
+}
+
+function PC_BOTH_WAIT_FOR_ICE_FAILED(test) {
+  var isFail = (f, reason, msg) =>
+    f().then(() => { throw new Error(msg + " must fail"); },
+             e => is(e.message, reason, msg + " must fail with: " + e.message));
+
+  return Promise.all([
+    isFail(() => test.pcLocal.waitForIceConnected(), "ICE failed", "Local ICE"),
+    isFail(() => test.pcRemote.waitForIceConnected(), "ICE failed", "Remote ICE")
+  ])
+  .then(() => ok(true, "ICE on both sides must fail."));
+}
+
+runNetworkTest(async options => {
+  await pushPrefs(
+    ['media.peerconnection.ice.stun_client_maximum_transmits', 3],
+    ['media.peerconnection.ice.trickle_grace_period', 5000]
+  );
+  options = options || {};
+  options.config_local = options.config_local || {};
+  const servers = options.config_local.iceServers || [];
+  // remove any turn servers
+  options.config_local.iceServers = servers.filter(server =>
+      server.urls.every(u => !u.toLowerCase().startsWith('turn')));
+
+  // Here's the setting we're testing. Comment out and this test should fail:
+  options.config_local.iceTransportPolicy = "relay";
+
+  const test = new PeerConnectionTest(options);
+  test.setMediaConstraints([{audio: true}, {video: true}],
+                           [{audio: true}, {video: true}]);
+  test.chain.remove("PC_LOCAL_SETUP_ICE_LOGGER");  // Needed to suppress failing
+  test.chain.remove("PC_REMOTE_SETUP_ICE_LOGGER"); // on ICE-failure.
+  test.chain.insertAfter("PC_LOCAL_SETUP_ICE_HANDLER", PC_LOCAL_NO_CANDIDATES);
+  test.chain.replace("PC_LOCAL_WAIT_FOR_ICE_CONNECTED", PC_BOTH_WAIT_FOR_ICE_FAILED);
+  test.chain.removeAfter("PC_BOTH_WAIT_FOR_ICE_FAILED");
+  await test.run();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_remoteReofferRollback.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_remoteReofferRollback.html
new file mode 100644
index 0000000000..80aa30beaa
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_remoteReofferRollback.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "952145",
+    title: "Rollback remote reoffer"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}, {audio: true}],
+                                   [{audio: true}]);
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+      ]
+    );
+    test.chain.replaceAfter('PC_REMOTE_SET_REMOTE_DESCRIPTION',
+      [
+        function PC_REMOTE_ROLLBACK(test) {
+          return test.setRemoteDescription(test.pcRemote, { type: "rollback" },
+                                           STABLE);
+        },
+
+        function PC_LOCAL_ROLLBACK(test) {
+          // We haven't negotiated the new stream yet.
+          test.pcLocal.expectNegotiationNeeded();
+          return test.setLocalDescription(
+              test.pcLocal,
+              new RTCSessionDescription({ type: "rollback", sdp: ""}),
+              STABLE);
+        },
+      ],
+      1 // Second PC_REMOTE_SET_REMOTE_DESCRIPTION
+    );
+    test.chain.append(commandsPeerConnectionOfferAnswer);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_remoteRollback.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_remoteRollback.html
new file mode 100644
index 0000000000..827646b0de
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_remoteRollback.html
@@ -0,0 +1,51 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "952145",
+    title: "Rollback remote offer"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    test.chain.removeAfter('PC_REMOTE_CHECK_CAN_TRICKLE_SYNC');
+    test.chain.append([
+      function PC_REMOTE_ROLLBACK(test) {
+        // We still haven't negotiated the tracks
+        test.pcRemote.expectNegotiationNeeded();
+        return test.setRemoteDescription(test.pcRemote, { type: "rollback" },
+                                         STABLE);
+      },
+
+      function PC_REMOTE_CHECK_CAN_TRICKLE_REVERT_SYNC(test) {
+        is(test.pcRemote._pc.canTrickleIceCandidates, null,
+           "Remote canTrickleIceCandidates is reverted to null");
+      },
+
+      function PC_LOCAL_ROLLBACK(test) {
+        // We still haven't negotiated the tracks
+        test.pcLocal.expectNegotiationNeeded();
+        return test.setLocalDescription(
+          test.pcLocal,
+          new RTCSessionDescription({ type: "rollback", sdp: ""}),
+          STABLE);
+      },
+
+      // Rolling back should shut down gathering
+      function PC_LOCAL_WAIT_FOR_END_OF_TRICKLE(test) {
+        return test.pcLocal.endOfTrickleIce;
+      },
+    ]);
+    test.chain.append(commandsPeerConnectionOfferAnswer);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_removeAudioTrack.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeAudioTrack.html
new file mode 100644
index 0000000000..e1e99b38c9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeAudioTrack.html
@@ -0,0 +1,57 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: remove audio track"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    let receivedTrack, analyser, freq;
+    addRenegotiation(test.chain,
+      [
+        function PC_REMOTE_SETUP_ANALYSER(test) {
+          is(test.pcRemote._pc.getReceivers().length, 1,
+             "pcRemote should have one receiver before renegotiation");
+
+          receivedTrack = test.pcRemote._pc.getReceivers()[0].track;
+          is(receivedTrack.readyState, "live",
+             "The received track should be live");
+
+          analyser = new AudioStreamAnalyser(
+              new AudioContext(), new MediaStream([receivedTrack]));
+          freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+
+          return analyser.waitForAnalysisSuccess(arr => arr[freq] > 200);
+        },
+        function PC_LOCAL_REMOVE_AUDIO_TRACK(test) {
+          test.setOfferOptions({ offerToReceiveAudio: true });
+          return test.pcLocal.removeSender(0);
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_FLOW_STOPPED(test) {
+          // Simply removing a track is not enough to cause it to be
+          // signaled as ended. Spec may change though.
+          // TODO: One last check of the spec is in order
+          is(receivedTrack.readyState, "live",
+             "The received track should not have ended");
+
+          return analyser.waitForAnalysisSuccess(arr => arr[freq] < 50);
+        },
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddAudioTrack.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddAudioTrack.html
new file mode 100644
index 0000000000..28b76e3b43
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddAudioTrack.html
@@ -0,0 +1,87 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: remove then add audio track"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    let originalTrack;
+    let haveMuteEvent = new Promise(() => {});
+    let haveUnmuteEvent = new Promise(() => {});
+    addRenegotiation(test.chain,
+      [
+        function PC_REMOTE_FIND_RECEIVER(test) {
+          is(test.pcRemote._pc.getReceivers().length, 1,
+             "pcRemote should have one receiver");
+          originalTrack = test.pcRemote._pc.getReceivers()[0].track;
+        },
+        function PC_LOCAL_REMOVE_AUDIO_TRACK(test) {
+          return test.pcLocal.removeSender(0);
+        },
+        function PC_LOCAL_ADD_AUDIO_TRACK(test) {
+          // The new track's pipeline will start with a packet count of
+          // 0, but the remote side will keep its old pipeline and packet
+          // count.
+          test.pcLocal.disableRtpCountChecking = true;
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+      ],
+      [
+        function PC_REMOTE_WAIT_FOR_UNMUTE() {
+          return haveUnmuteEvent;
+        },
+        function PC_REMOTE_CHECK_ADDED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[1].receiver.track;
+
+          const analyser = new AudioStreamAnalyser(
+              new AudioContext(), new MediaStream([track]));
+          const freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+          return analyser.waitForAnalysisSuccess(arr => arr[freq] > 200);
+        },
+        function PC_REMOTE_WAIT_FOR_MUTE() {
+          return haveMuteEvent;
+        },
+        function PC_REMOTE_CHECK_REMOVED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[0].receiver.track;
+
+          const analyser = new AudioStreamAnalyser(
+              new AudioContext(), new MediaStream([track]));
+          const freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+          return analyser.waitForAnalysisSuccess(arr => arr[freq] < 50);
+        }
+      ]
+    );
+
+    // The first track should mute when the connection is closed.
+    test.chain.insertBefore("PC_REMOTE_SET_REMOTE_DESCRIPTION", [
+      function PC_REMOTE_SETUP_ONMUTE(test) {
+        haveMuteEvent = haveEvent(test.pcRemote._pc.getReceivers()[0].track, "mute");
+      }
+    ]);
+
+    // Second negotiation should cause the second track to unmute.
+    test.chain.insertAfter("PC_REMOTE_SET_REMOTE_DESCRIPTION", [
+      function PC_REMOTE_SETUP_ONUNMUTE(test) {
+        haveUnmuteEvent = haveEvent(test.pcRemote._pc.getReceivers()[1].track, "unmute");
+      }
+    ], false, 1);
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddAudioTrackNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddAudioTrackNoBundle.html
new file mode 100644
index 0000000000..cff424e12c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddAudioTrackNoBundle.html
@@ -0,0 +1,76 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: remove then add audio track"
+  });
+
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.bundle = false;
+    const test = new PeerConnectionTest(options);
+    let originalTrack;
+    addRenegotiation(test.chain,
+      [
+        function PC_REMOTE_FIND_RECEIVER(test) {
+          is(test.pcRemote._pc.getReceivers().length, 1,
+             "pcRemote should have one receiver");
+          originalTrack = test.pcRemote._pc.getReceivers()[0].track;
+        },
+        function PC_LOCAL_REMOVE_AUDIO_TRACK(test) {
+          // The new track's pipeline will start with a packet count of
+          // 0, but the remote side will keep its old pipeline and packet
+          // count.
+          test.pcLocal.disableRtpCountChecking = true;
+          return test.pcLocal.removeSender(0);
+        },
+        function PC_LOCAL_ADD_AUDIO_TRACK(test) {
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_ADDED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[1].receiver.track;
+
+          const analyser = new AudioStreamAnalyser(
+              new AudioContext(), new MediaStream([track]));
+          const freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+          return analyser.waitForAnalysisSuccess(arr => arr[freq] > 200);
+        },
+        function PC_REMOTE_CHECK_REMOVED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[0].receiver.track;
+
+          const analyser = new AudioStreamAnalyser(
+              new AudioContext(), new MediaStream([track]));
+          const freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+          return analyser.waitForAnalysisSuccess(arr => arr[freq] < 50);
+        }
+      ]
+    );
+
+    test.chain.insertAfterEach('PC_LOCAL_CREATE_OFFER',
+                               PC_LOCAL_REMOVE_BUNDLE_FROM_OFFER);
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddVideoTrack.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddVideoTrack.html
new file mode 100644
index 0000000000..b1be690e5b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddVideoTrack.html
@@ -0,0 +1,98 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: remove then add video track"
+  });
+
+  runNetworkTest(async function (options) {
+    // Use fake video here since the native fake device on linux doesn't
+    // change color as needed by checkVideoPlaying() below.
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    const test = new PeerConnectionTest(options);
+    const helper = new VideoStreamHelper();
+    var originalTrack;
+    let haveMuteEvent = new Promise(() => {});
+    let haveUnmuteEvent = new Promise(() => {});
+    addRenegotiation(test.chain,
+      [
+        function PC_REMOTE_FIND_RECEIVER(test) {
+          is(test.pcRemote._pc.getReceivers().length, 1,
+             "pcRemote should have one receiver");
+          originalTrack = test.pcRemote._pc.getReceivers()[0].track;
+        },
+        function PC_LOCAL_REMOVE_VIDEO_TRACK(test) {
+          // The new track's pipeline will start with a packet count of
+          // 0, but the remote side will keep its old pipeline and packet
+          // count.
+          test.pcLocal.disableRtpCountChecking = true;
+          return test.pcLocal.removeSender(0);
+        },
+        function PC_LOCAL_ADD_VIDEO_TRACK(test) {
+          return test.pcLocal.getAllUserMediaAndAddStreams([{video: true}]);
+        },
+      ],
+      [
+        function PC_REMOTE_WAIT_FOR_UNMUTE() {
+          return haveUnmuteEvent;
+        },
+        function PC_REMOTE_CHECK_ADDED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[1].receiver.track;
+
+          const vAdded = test.pcRemote.remoteMediaElements.find(
+              elem => elem.id.includes(track.id));
+          return helper.checkVideoPlaying(vAdded);
+        },
+        function PC_REMOTE_WAIT_FOR_MUTE() {
+          return haveMuteEvent;
+        },
+        function PC_REMOTE_CHECK_REMOVED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[0].receiver.track;
+
+          const vAdded = test.pcRemote.remoteMediaElements.find(
+              elem => elem.id.includes(track.id));
+          return helper.checkVideoPaused(vAdded, 10, 10, 16, 5000);
+        }
+      ]
+    );
+
+    // The first track should mute when the connection is closed.
+    test.chain.insertBefore("PC_REMOTE_SET_REMOTE_DESCRIPTION", [
+      function PC_REMOTE_SETUP_ONMUTE(test) {
+        haveMuteEvent = haveEvent(test.pcRemote._pc.getReceivers()[0].track, "mute");
+      }
+    ]);
+
+    // Second negotiation should cause the second track to unmute.
+    test.chain.insertAfter("PC_REMOTE_SET_REMOTE_DESCRIPTION", [
+      function PC_REMOTE_SETUP_ONUNMUTE(test) {
+        haveUnmuteEvent = haveEvent(test.pcRemote._pc.getReceivers()[1].track, "unmute");
+      }
+    ], false, 1);
+
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddVideoTrackNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddVideoTrackNoBundle.html
new file mode 100644
index 0000000000..dcaf7943e2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddVideoTrackNoBundle.html
@@ -0,0 +1,89 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: remove then add video track, no bundle"
+  });
+
+  runNetworkTest(async function (options) {
+    // Use fake video here since the native fake device on linux doesn't
+    // change color as needed by checkVideoPlaying() below.
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    options = options || { };
+    options.bundle = false;
+    const test = new PeerConnectionTest(options);
+    const helper = new VideoStreamHelper();
+    var originalTrack;
+    addRenegotiation(test.chain,
+      [
+        function PC_REMOTE_FIND_RECEIVER(test) {
+          is(test.pcRemote._pc.getReceivers().length, 1,
+             "pcRemote should have one receiver");
+          originalTrack = test.pcRemote._pc.getReceivers()[0].track;
+        },
+        function PC_LOCAL_REMOVE_VIDEO_TRACK(test) {
+          // The new track's pipeline will start with a packet count of
+          // 0, but the remote side will keep its old pipeline and packet
+          // count.
+          test.pcLocal.disableRtpCountChecking = true;
+          return test.pcLocal.removeSender(0);
+        },
+        function PC_LOCAL_ADD_VIDEO_TRACK(test) {
+          // Use fake:true here since the native fake device on linux doesn't
+          // change color as needed by checkVideoPlaying() below.
+          return test.pcLocal.getAllUserMediaAndAddStreams([{video: true}]);
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_ADDED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[1].receiver.track;
+
+          const vAdded = test.pcRemote.remoteMediaElements.find(
+              elem => elem.id.includes(track.id));
+          return helper.checkVideoPlaying(vAdded);
+        },
+        function PC_REMOTE_CHECK_REMOVED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[0].receiver.track;
+
+          const vAdded = test.pcRemote.remoteMediaElements.find(
+              elem => elem.id.includes(track.id));
+          return helper.checkVideoPaused(vAdded, 10, 10, 16, 5000);
+        },
+      ]
+    );
+
+    test.chain.insertAfterEach('PC_LOCAL_CREATE_OFFER',
+                               PC_LOCAL_REMOVE_BUNDLE_FROM_OFFER);
+
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_removeVideoTrack.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeVideoTrack.html
new file mode 100644
index 0000000000..4c4e7905e1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeVideoTrack.html
@@ -0,0 +1,64 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: remove video track"
+  });
+
+  runNetworkTest(async (options) => {
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    const test = new PeerConnectionTest(options);
+    let receivedTrack, element;
+    addRenegotiation(test.chain,
+      [
+        function PC_REMOTE_SETUP_HELPER(test) {
+          is(test.pcRemote._pc.getReceivers().length, 1,
+             "pcRemote should have one receiver before renegotiation");
+
+          receivedTrack = test.pcRemote._pc.getReceivers()[0].track;
+          is(receivedTrack.readyState, "live",
+             "The received track should be live");
+
+          element = createMediaElement("video", "pcRemoteReceivedVideo");
+          element.srcObject = new MediaStream([receivedTrack]);
+          return haveEvent(element, "loadeddata");
+        },
+        function PC_LOCAL_REMOVE_VIDEO_TRACK(test) {
+          test.setOfferOptions({ offerToReceiveVideo: true });
+          test.setMediaConstraints([], [{video: true}]);
+          return test.pcLocal.removeSender(0);
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_FLOW_STOPPED(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 1,
+              "pcRemote should have one transceiver");
+          const track = test.pcRemote._pc.getTransceivers()[0].receiver.track;
+
+          const vAdded = test.pcRemote.remoteMediaElements.find(
+              elem => elem.id.includes(track.id));
+          const helper = new VideoStreamHelper();
+          return helper.checkVideoPaused(vAdded, 10, 10, 16, 5000);
+        },
+      ]
+    );
+
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_renderAfterRenegotiation.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_renderAfterRenegotiation.html
new file mode 100644
index 0000000000..c8091d7a9e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_renderAfterRenegotiation.html
@@ -0,0 +1,89 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1273652",
+    title: "Video receiver still renders after renegotiation",
+    visible: true
+  });
+
+  var pc1 = new RTCPeerConnection();
+  var pc2 = new RTCPeerConnection();
+
+  var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+  pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
+  pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+  var v1, v2;
+  var delivered = new Promise(resolve => pc2.ontrack = e => {
+    // Test RTCTrackEvent here.
+    ok(e.streams.length, "has streams");
+    ok(e.streams[0].getTrackById(e.track.id), "has track");
+    ok(pc2.getReceivers().some(receiver => receiver == e.receiver), "has receiver");
+    if (e.streams[0].getTracks().length == 1) {
+      // Test RTCTrackEvent required args here.
+      mustThrowWith("RTCTrackEvent wo/required args",
+                    "TypeError", () => new RTCTrackEvent("track", {}));
+      v2.srcObject = e.streams[0];
+      resolve();
+    }
+  });
+
+  runNetworkTest(async () => {
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false]);
+      await pushPrefs(["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    v2 = createMediaElement('video', 'v2');
+    is(v2.currentTime, 0, "v2.currentTime is zero at outset");
+
+    const emitter = new VideoFrameEmitter(CaptureStreamTestHelper.prototype.blue,
+                                          CaptureStreamTestHelper.prototype.green,
+                                          16, 16);
+    emitter.start();
+    emitter.stream().getTracks().forEach(t => pc1.addTrack(t, emitter.stream()));
+    let h = emitter.helper();
+
+    let offer = await pc1.createOffer({});
+    await pc1.setLocalDescription(offer);
+    await pc2.setRemoteDescription(pc1.localDescription);
+    // check that createAnswer accepts arg.
+    let answer = await pc2.createAnswer({});
+    await pc2.setLocalDescription(answer);
+    await pc1.setRemoteDescription(pc2.localDescription);
+
+    // re-negotiate to trigger the race condition in the jitter buffer
+    offer = await pc1.createOffer({}); // check that createOffer accepts arg.
+    await pc1.setLocalDescription(offer);
+    await pc2.setRemoteDescription(pc1.localDescription);
+    answer = await pc2.createAnswer({});
+    await pc2.setLocalDescription(answer);
+    await pc1.setRemoteDescription(pc2.localDescription);
+    await delivered;
+
+    // now verify that actually something gets rendered into the remote video
+    // element.
+    await h.pixelMustBecome(v2, h.blue, {
+      threshold: 128,
+      infoString: "pcRemote's video should become blue",
+    });
+    // This will verify that new changes to the canvas propagate through
+    // the peerconnection
+    emitter.colors(h.red, h.green)
+    await h.pixelMustBecome(v2, h.red, {
+      threshold: 128,
+      infoString: "pcRemote's video should become red",
+    });
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceNullTrackThenRenegotiateAudio.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceNullTrackThenRenegotiateAudio.html
new file mode 100644
index 0000000000..2253b87672
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceNullTrackThenRenegotiateAudio.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1763832",
+    title: "Renegotiation (audio): Start with no track and recvonly, then replace and set direction to sendrecv, then renegotiate"
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+      ['media.audio_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+    const transceiverSend = offerer.addTransceiver('audio', {direction: 'recvonly'});
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    await offerer.setLocalDescription();
+    await answerer.setRemoteDescription(offerer.localDescription);
+    await answerer.setLocalDescription();
+    await offerer.setRemoteDescription(answerer.localDescription);
+
+    // add audio with replaceTrack, set send bit, and renegotiate
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    const [track] = stream.getAudioTracks();
+    transceiverSend.sender.replaceTrack(track);
+    transceiverSend.direction = "sendrecv";
+    const remoteStreamAvailable = new Promise(r => {
+      answerer.ontrack = ({track}) => r(new MediaStream([track]));
+    });
+
+    await offerer.setLocalDescription();
+    await answerer.setRemoteDescription(offerer.localDescription);
+    await answerer.setLocalDescription();
+    await offerer.setRemoteDescription(answerer.localDescription);
+
+    const remoteStream = await remoteStreamAvailable;
+    const h = new AudioStreamHelper();
+    await h.checkAudioFlowing(remoteStream);
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceNullTrackThenRenegotiateVideo.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceNullTrackThenRenegotiateVideo.html
new file mode 100644
index 0000000000..d7bd6d8a37
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceNullTrackThenRenegotiateVideo.html
@@ -0,0 +1,63 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script></head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1763832",
+    title: "Renegotiation (video): Start with no track and recvonly, then replace and set direction to sendrecv, then renegotiate"
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+    const transceiverSend = offerer.addTransceiver('video', {direction: 'recvonly'});
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    await offerer.setLocalDescription();
+    await answerer.setRemoteDescription(offerer.localDescription);
+    await answerer.setLocalDescription();
+    await offerer.setRemoteDescription(answerer.localDescription);
+
+    // add video with replaceTrack, set send bit, and renegotiate
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const [track] = stream.getVideoTracks();
+    transceiverSend.sender.replaceTrack(track);
+    transceiverSend.direction = "sendrecv";
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    await offerer.setLocalDescription();
+    await answerer.setRemoteDescription(offerer.localDescription);
+    await answerer.setLocalDescription();
+    await offerer.setRemoteDescription(answerer.localDescription);
+
+    const elems = await Promise.all(metadataToBeLoaded);
+    is(elems.length, 1, "Should have one video element");
+
+    const helper = new VideoStreamHelper();
+    await helper.checkVideoPlaying(elems[0]);
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack.html
new file mode 100644
index 0000000000..9befc5c564
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack.html
@@ -0,0 +1,187 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1032839",
+    title: "Replace video and audio (with WebAudio) tracks",
+    visible: true
+  });
+
+  function allLocalStreamsHaveSender(pc) {
+    return pc.getLocalStreams()
+      .every(s => s.getTracks()        // Every local stream,
+        .some(t => pc.getSenders()     // should have some track,
+          .some(sn => sn.track == t))) // that's being sent over |pc|.
+  }
+
+  function allRemoteStreamsHaveReceiver(pc) {
+    return pc.getRemoteStreams()
+      .every(s => s.getTracks()        // Every remote stream,
+        .some(t => pc.getReceivers()   // should have some track,
+          .some(sn => sn.track == t))) // that's being received over |pc|.
+  }
+
+  function replacetest(wrapper) {
+    var pc = wrapper._pc;
+    var oldSenderCount = pc.getSenders().length;
+    var sender = pc.getSenders().find(sn => sn.track.kind == "video");
+    var oldTrack = sender.track;
+    ok(sender, "We have a sender for video");
+    ok(allLocalStreamsHaveSender(pc),
+       "Shouldn't have any local streams without a corresponding sender");
+    ok(allRemoteStreamsHaveReceiver(pc),
+       "Shouldn't have any remote streams without a corresponding receiver");
+
+    var newTrack;
+    var audiotrack;
+    return getUserMedia({video:true, audio:true})
+      .then(newStream => {
+        window.grip = newStream;
+        newTrack = newStream.getVideoTracks()[0];
+        audiotrack = newStream.getAudioTracks()[0];
+        isnot(newTrack, sender.track, "replacing with a different track");
+        ok(!pc.getLocalStreams().some(s => s == newStream),
+           "from a different stream");
+        // Use wrapper function, since it updates expected tracks
+        return wrapper.senderReplaceTrack(sender, newTrack, newStream);
+      })
+      .then(() => {
+        is(pc.getSenders().length, oldSenderCount, "same sender count");
+        is(sender.track, newTrack, "sender.track has been replaced");
+        ok(!pc.getSenders().map(sn => sn.track).some(t => t == oldTrack),
+           "old track not among senders");
+        // Spec does not say we add this new track to any stream
+        ok(!pc.getLocalStreams().some(s => s.getTracks()
+                                           .some(t => t == sender.track)),
+           "track does not exist among pc's local streams");
+        return sender.replaceTrack(audiotrack)
+          .then(() => ok(false, "replacing with different kind should fail"),
+                e => is(e.name, "TypeError",
+                        "replacing with different kind should fail"));
+      });
+  }
+
+  runNetworkTest(function () {
+    test = new PeerConnectionTest();
+    test.audioCtx = new AudioContext();
+    test.setMediaConstraints([{video: true, audio: true}], [{video: true}]);
+    test.chain.removeAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW");
+
+    // Test replaceTrack on pcRemote separately since it's video only.
+    test.chain.append([
+      function PC_REMOTE_VIDEOONLY_REPLACE_VIDEOTRACK(test) {
+        return replacetest(test.pcRemote);
+      },
+      function PC_LOCAL_NEW_VIDEOTRACK_WAIT_FOR_MEDIA_FLOW(test) {
+        return test.pcLocal.waitForMediaFlow();
+      }
+    ]);
+
+    // Replace video twice on pcLocal to make sure it still works
+    // (does audio twice too, but hey)
+    test.chain.append([
+      function PC_LOCAL_AUDIOVIDEO_REPLACE_VIDEOTRACK_1(test) {
+        return replacetest(test.pcLocal);
+      },
+      function PC_REMOTE_NEW_VIDEOTRACK_WAIT_FOR_MEDIA_FLOW_1(test) {
+        return test.pcRemote.waitForMediaFlow();
+      },
+      function PC_LOCAL_AUDIOVIDEO_REPLACE_VIDEOTRACK_2(test) {
+        return replacetest(test.pcLocal);
+      },
+      function PC_REMOTE_NEW_VIDEOTRACK_WAIT_FOR_MEDIA_FLOW_2(test) {
+        return test.pcRemote.waitForMediaFlow();
+      }
+    ]);
+
+    test.chain.append([
+      function PC_LOCAL_AUDIOVIDEO_REPLACE_VIDEOTRACK_WITHSAME(test) {
+        var pc = test.pcLocal._pc;
+        var sender = pc.getSenders().find(sn => sn.track.kind == "video");
+        ok(sender, "should still have a sender of video");
+        return sender.replaceTrack(sender.track)
+          .then(() => ok(true, "replacing with itself should succeed"));
+      },
+      function PC_REMOTE_NEW_SAME_VIDEOTRACK_WAIT_FOR_MEDIA_FLOW(test) {
+        return test.pcRemote.waitForMediaFlow();
+      }
+    ]);
+
+    // Replace the gUM audio track on pcLocal with a WebAudio track.
+    test.chain.append([
+      function PC_LOCAL_AUDIOVIDEO_REPLACE_AUDIOTRACK_WEBAUDIO(test) {
+        var pc = test.pcLocal._pc;
+        var sender = pc.getSenders().find(sn => sn.track.kind == "audio");
+        ok(sender, "track has a sender");
+        var oldSenderCount = pc.getSenders().length;
+        var oldTrack = sender.track;
+
+        var sourceNode = test.audioCtx.createOscillator();
+        sourceNode.type = 'sine';
+        // We need a frequency not too close to the fake audio track
+        // (440Hz for loopback devices, 1kHz for fake tracks).
+        sourceNode.frequency.value = 2000;
+        sourceNode.start();
+
+        var destNode = test.audioCtx.createMediaStreamDestination();
+        sourceNode.connect(destNode);
+        var newTrack = destNode.stream.getAudioTracks()[0];
+
+        return test.pcLocal.senderReplaceTrack(
+            sender, newTrack, destNode.stream)
+          .then(() => {
+            is(pc.getSenders().length, oldSenderCount, "same sender count");
+            ok(!pc.getSenders().some(sn => sn.track == oldTrack),
+               "Replaced track should be removed from senders");
+            // TODO: Should PC remove local streams when there are no senders
+            // associated with it? getLocalStreams() isn't in the spec anymore,
+            // so I guess it is pretty arbitrary?
+            is(sender.track, newTrack, "sender.track has been replaced");
+            // Spec does not say we add this new track to any stream
+            ok(!pc.getLocalStreams().some(s => s.getTracks()
+                                               .some(t => t == sender.track)),
+               "track exists among pc's local streams");
+          });
+      }
+    ]);
+    test.chain.append([
+      function PC_LOCAL_CHECK_WEBAUDIO_FLOW_PRESENT(test) {
+        return test.pcRemote.checkReceivingToneFrom(test.audioCtx, test.pcLocal);
+      }
+    ]);
+    test.chain.append([
+      function PC_LOCAL_INVALID_ADD_VIDEOTRACKS(test) {
+        let videoTransceivers = test.pcLocal._pc.getTransceivers()
+          .filter(transceiver => {
+            return !transceiver.stopped &&
+                   transceiver.receiver.track.kind == "video" &&
+                   transceiver.sender.track;
+          });
+
+        ok(videoTransceivers.length,
+           "There is at least one non-stopped video transceiver with a track.");
+
+        videoTransceivers.forEach(transceiver => {
+            var stream = test.pcLocal._pc.getLocalStreams()[0];;
+            var track = transceiver.sender.track;
+            try {
+              test.pcLocal._pc.addTrack(track, stream);
+              ok(false, "addTrack existing track should fail");
+            } catch (e) {
+              is(e.name, "InvalidAccessError",
+                 "addTrack existing track should fail");
+            }
+          });
+      }
+    ]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_camera.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_camera.html
new file mode 100644
index 0000000000..356517e79f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_camera.html
@@ -0,0 +1,48 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+
+<video id="video" width="160" height="120" autoplay></video>
+
+<script type="application/javascript">
+  createHTML({
+    bug: "1709481",
+    title: "replaceTrack (null -> camera) test",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    // Make sure we use the fake video device, and not loopback
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    const pc1 = new RTCPeerConnection(), pc2 = new RTCPeerConnection();
+    pc1.onicecandidate = e => pc2.addIceCandidate(e.candidate);
+    pc2.onicecandidate = e => pc1.addIceCandidate(e.candidate);
+    pc2.ontrack = ({track}) => video.srcObject = new MediaStream([track]);
+    pc1.addTransceiver("audio");
+    const tc1 = pc1.addTransceiver("video");
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const [track] = stream.getVideoTracks();
+    await pc1.setLocalDescription();
+    await pc2.setRemoteDescription(pc1.localDescription);
+    await pc2.setLocalDescription();
+    await pc1.setRemoteDescription(pc2.localDescription);
+    await wait(100);
+    await tc1.sender.replaceTrack(track);
+    const h = new VideoStreamHelper();
+    await h.checkVideoPlaying(video);
+    pc1.close();
+    pc2.close();
+    await SpecialPowers.popPrefEnv();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_disabled.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_disabled.html
new file mode 100644
index 0000000000..11b2762d96
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_disabled.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+<script src="pc.js"></script>
+<script src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script>
+createHTML({
+  bug: "1576771",
+  title: "Replace a disabled video track with an enabled one",
+  visible: true,
+});
+
+runNetworkTest(() => {
+  const helper = new CaptureStreamTestHelper2D(240, 160);
+  const emitter = new VideoFrameEmitter(helper.green, helper.green, 240, 160);
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW", [
+    function PC_LOCAL_DISABLE_SENDTRACK(test) {
+      test.pcLocal._pc.getSenders()[0].track.enabled = false;
+    },
+    function PC_REMOTE_WAIT_FOR_BLACK(test) {
+      return helper.pixelMustBecome(
+        test.pcRemote.remoteMediaElements[0], helper.black, {
+          threshold: 128,
+          infoString: "Remote disabled track becomes black",
+          cancel: wait(10000).then(
+            () => new Error("Timeout waiting for black"))});
+    },
+    function PC_LOCAL_REPLACETRACK_WITH_ENABLED_TRACK(test) {
+      emitter.start();
+      test.pcLocal._pc.getSenders()[0].replaceTrack(
+        emitter.stream().getTracks()[0]);
+    },
+  ]);
+  test.chain.append([
+    function PC_REMOTE_WAIT_FOR_GREEN(test) {
+      return helper.pixelMustBecome(
+        test.pcRemote.remoteMediaElements[0], helper.green, {
+          threshold: 128,
+          infoString: "Remote disabled track becomes green",
+          cancel: wait(10000).then(
+            () => new Error("Timeout waiting for green"))});
+    },
+    function CLEANUP(test) {
+      emitter.stop();
+      for (const track of emitter.stream().getTracks()) {
+        track.stop();
+      }
+    },
+  ]);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_microphone.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_microphone.html
new file mode 100644
index 0000000000..5886caf6a4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_microphone.html
@@ -0,0 +1,46 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+
+<script type="application/javascript">
+  createHTML({
+    bug: "1709481",
+    title: "replaceTrack (null -> microphone) test",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    // Make sure we use the fake audio device, and not loopback
+    await pushPrefs(
+      ['media.audio_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    const pc1 = new RTCPeerConnection(), pc2 = new RTCPeerConnection();
+    pc1.onicecandidate = e => pc2.addIceCandidate(e.candidate);
+    pc2.onicecandidate = e => pc1.addIceCandidate(e.candidate);
+    let remoteStream;
+    pc2.ontrack = ({track}) => remoteStream = new MediaStream([track]);
+    const tc1 = pc1.addTransceiver("audio");
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    const [track] = stream.getAudioTracks();
+    await pc1.setLocalDescription();
+    await pc2.setRemoteDescription(pc1.localDescription);
+    await pc2.setLocalDescription();
+    await pc1.setRemoteDescription(pc2.localDescription);
+    await wait(100);
+    await tc1.sender.replaceTrack(track);
+    const h = new AudioStreamHelper();
+    await h.checkAudioFlowing(remoteStream);
+    pc1.close();
+    pc2.close();
+    await SpecialPowers.popPrefEnv();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceVideoThenRenegotiate.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceVideoThenRenegotiate.html
new file mode 100644
index 0000000000..070cb42fcb
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceVideoThenRenegotiate.html
@@ -0,0 +1,74 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: replaceTrack followed by adding a second video stream"
+  });
+
+  runNetworkTest(async (options) => {
+    await pushPrefs(['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video:true}], [{video:true}]);
+    const helper = new VideoStreamHelper();
+    const emitter1 = new VideoFrameEmitter(CaptureStreamTestHelper.prototype.red,
+                                           CaptureStreamTestHelper.prototype.green);
+    const emitter2 = new VideoFrameEmitter(CaptureStreamTestHelper.prototype.blue,
+                                           CaptureStreamTestHelper.prototype.grey);
+    test.chain.replace("PC_LOCAL_GUM", [
+      function PC_LOCAL_ADDTRACK(test) {
+        test.pcLocal.attachLocalStream(emitter1.stream());
+        emitter1.start();
+      },
+    ]);
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_REPLACE_VIDEO_TRACK_THEN_ADD_SECOND_STREAM(test) {
+          emitter1.stop();
+          emitter2.start();
+          const newstream = emitter2.stream();
+          const newtrack = newstream.getVideoTracks()[0];
+          var sender = test.pcLocal._pc.getSenders()[0];
+          return test.pcLocal.senderReplaceTrack(sender, newtrack, newstream)
+            .then(() => {
+              test.setMediaConstraints([{video: true}, {video: true}],
+                                       [{video: true}]);
+            });
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_ORIGINAL_TRACK_NOT_ENDED(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 1,
+              "pcRemote should have one transceiver");
+          const track = test.pcRemote._pc.getTransceivers()[0].receiver.track;
+
+          const vremote = test.pcRemote.remoteMediaElements.find(
+              elem => elem.id.includes(track.id));
+          if (!vremote) {
+            return Promise.reject(new Error("Couldn't find video element"));
+          }
+          ok(!vremote.ended, "Original track should not have ended after renegotiation (replaceTrack is not signalled!)");
+          return helper.checkVideoPlaying(vremote);
+        }
+      ]
+    );
+
+    await test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIce.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIce.html
new file mode 100644
index 0000000000..d94bb084b7
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIce.html
@@ -0,0 +1,41 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        }
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceBadAnswer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceBadAnswer.html
new file mode 100644
index 0000000000..b71001b0db
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceBadAnswer.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1413709",
+    title: "Renegotiation: bad answer ICE credentials"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}],
+                                   []);
+          return test.pcLocal.getAllUserMedia([{audio: true}]);
+        },
+      ]
+    );
+
+    // If the offerer hasn't indicated ICE restart, then an answer
+    // arriving during renegotiation that has modified ICE credentials
+    // should cause an error
+    test.chain.replaceAfter("PC_LOCAL_GET_ANSWER",
+      [
+        function PC_LOCAL_REWRITE_REMOTE_SDP_ICE_CREDS(test) {
+          test._remote_answer.sdp =
+            test._remote_answer.sdp.replace(/a=ice-pwd:.*\r\n/g,
+                                            "a=ice-pwd:bad-pwd\r\n")
+                                   .replace(/a=ice-ufrag:.*\r\n/g,
+                                            "a=ice-ufrag:bad-ufrag\r\n");
+        },
+
+        function PC_LOCAL_EXPECT_SET_REMOTE_DESCRIPTION_FAIL(test) {
+          return test.setRemoteDescription(test.pcLocal,
+                                           test._remote_answer,
+                                           STABLE)
+           .then(() => ok(false, "setRemoteDescription must fail"),
+                 e => is(e.name, "InvalidAccessError",
+                         "setRemoteDescription must fail and did"));
+         }
+      ], 1 // replace after the second PC_LOCAL_GET_ANSWER
+    );
+
+    test.setMediaConstraints([{audio: true}], []);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalAndRemoteRollback.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalAndRemoteRollback.html
new file mode 100644
index 0000000000..6bbf9440fc
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalAndRemoteRollback.html
@@ -0,0 +1,82 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, local and remote rollback"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        async function PC_LOCAL_SETUP_ICE_HANDLER(test) {
+          await test.pcLocal.endOfTrickleIce;
+          test.pcLocal.setupIceCandidateHandler(test);
+        },
+
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        }
+      ]
+    );
+
+    test.chain.replaceAfter('PC_REMOTE_CREATE_ANSWER',
+      [
+        function PC_LOCAL_EXPECT_ICE_CONNECTED(test) {
+          test.pcLocal.iceCheckingIceRollbackExpected = true;
+        },
+
+        function PC_REMOTE_ROLLBACK(test) {
+          return test.setRemoteDescription(test.pcRemote, { type: "rollback" },
+                                           STABLE);
+        },
+
+        async function PC_LOCAL_ROLLBACK(test) {
+          await test.pcLocal.endOfTrickleIce;
+          // We haven't negotiated the new stream yet.
+          test.pcLocal.expectNegotiationNeeded();
+          return test.setLocalDescription(
+              test.pcLocal,
+              new RTCSessionDescription({ type: "rollback", sdp: ""}),
+              STABLE);
+        },
+
+        // Rolling back should shut down gathering for the offerer,
+        // but because the answerer never set a local description, no ICE
+        // gathering has happened yet, so there's no changes to ICE gathering
+        // state
+        function PC_LOCAL_WAIT_FOR_END_OF_TRICKLE(test) {
+          return test.pcLocal.endOfTrickleIce;
+        },
+
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        }
+      ],
+      1 // Replaces after second PC_REMOTE_CREATE_ANSWER
+    );
+    test.chain.append(commandsPeerConnectionOfferAnswer);
+
+    // for now, only use one stream, because rollback doesn't seem to
+    // like multiple streams.  See bug 1259465.
+    test.setMediaConstraints([{audio: true}],
+                             [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalAndRemoteRollbackNoSubsequentRestart.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalAndRemoteRollbackNoSubsequentRestart.html
new file mode 100644
index 0000000000..37b0fc68fc
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalAndRemoteRollbackNoSubsequentRestart.html
@@ -0,0 +1,77 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, local and remote rollback, without a subsequent ICE restart"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_SETUP_ICE_HANDLER(test) {
+          test.pcLocal.setupIceCandidateHandler(test);
+        },
+        function PC_REMOTE_SETUP_ICE_HANDLER(test) {
+          test.pcRemote.setupIceCandidateHandler(test);
+        },
+
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        }
+      ]
+    );
+
+    test.chain.replaceAfter('PC_REMOTE_CREATE_ANSWER',
+      [
+        function PC_REMOTE_ROLLBACK(test) {
+          return test.setRemoteDescription(test.pcRemote, { type: "rollback" },
+                                           STABLE);
+        },
+
+        async function PC_LOCAL_ROLLBACK(test) {
+          await test.pcLocal.endOfTrickleIce;
+          // We haven't negotiated the new stream yet.
+          test.pcLocal.expectNegotiationNeeded();
+          return test.setLocalDescription(
+              test.pcLocal,
+              new RTCSessionDescription({ type: "rollback", sdp: ""}),
+              STABLE);
+        },
+
+        // Rolling back should shut down gathering for the offerer,
+        // but because the answerer never set a local description, no ICE
+        // gathering has happened yet, so there's no changes to ICE gathering
+        // state
+        function PC_LOCAL_WAIT_FOR_END_OF_TRICKLE(test) {
+          return test.pcLocal.endOfTrickleIce;
+        },
+
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: false });
+        }
+      ],
+      1 // Replaces after second PC_REMOTE_CREATE_ANSWER
+    );
+    test.chain.append(commandsPeerConnectionOfferAnswer);
+
+    // for now, only use one stream, because rollback doesn't seem to
+    // like multiple streams.  See bug 1259465.
+    test.setMediaConstraints([{audio: true}],
+                             [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalRollback.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalRollback.html
new file mode 100644
index 0000000000..f5f9a1f220
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalRollback.html
@@ -0,0 +1,76 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, local rollback"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        },
+        // causes an ice restart and then rolls it back
+        // (does not result in sending an offer)
+        function PC_LOCAL_SETUP_ICE_HANDLER(test) {
+          test.pcLocal.setupIceCandidateHandler(test, () => {});
+        },
+        function PC_LOCAL_CREATE_AND_SET_OFFER(test) {
+          return test.createOffer(test.pcLocal).then(offer => {
+            return test.setLocalDescription(test.pcLocal,
+                                            offer,
+                                            HAVE_LOCAL_OFFER);
+          });
+        },
+        function PC_LOCAL_EXPECT_ICE_CONNECTED(test) {
+          test.pcLocal.iceCheckingIceRollbackExpected = true;
+        },
+        function PC_LOCAL_WAIT_FOR_GATHERING(test) {
+          return new Promise(r => {
+            test.pcLocal._pc.addEventListener("icegatheringstatechange", () => {
+              if (test.pcLocal._pc.iceGatheringState == "gathering") {
+                r();
+              }
+            });
+          });
+        },
+        function PC_LOCAL_ROLLBACK(test) {
+          return test.setLocalDescription(test.pcLocal,
+                                          { type: "rollback", sdp: ""},
+                                          STABLE);
+        },
+        // Rolling back should shut down gathering
+        function PC_LOCAL_WAIT_FOR_END_OF_TRICKLE(test) {
+          return test.pcLocal.endOfTrickleIce;
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        }
+      ]
+    );
+
+    // for now, only use one stream, because rollback doesn't seem to
+    // like multiple streams.  See bug 1259465.
+    test.setMediaConstraints([{audio: true}],
+                             [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalRollbackNoSubsequentRestart.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalRollbackNoSubsequentRestart.html
new file mode 100644
index 0000000000..8e27864aae
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalRollbackNoSubsequentRestart.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, local rollback, then renegotiation without ICE restart"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        },
+        // causes an ice restart and then rolls it back
+        // (does not result in sending an offer)
+        function PC_LOCAL_SETUP_ICE_HANDLER(test) {
+          test.pcLocal.setupIceCandidateHandler(test, () => {});
+        },
+        function PC_LOCAL_CREATE_AND_SET_OFFER(test) {
+          return test.createOffer(test.pcLocal).then(offer => {
+            return test.setLocalDescription(test.pcLocal,
+                                            offer,
+                                            HAVE_LOCAL_OFFER);
+          });
+        },
+        function PC_LOCAL_WAIT_FOR_END_OF_TRICKLE(test) {
+          return test.pcLocal.endOfTrickleIce;
+        },
+        function PC_LOCAL_ROLLBACK(test) {
+          return test.setLocalDescription(test.pcLocal,
+                                          { type: "rollback", sdp: ""},
+                                          STABLE);
+        },
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: false });
+        }
+      ]
+    );
+
+    // for now, only use one stream, because rollback doesn't seem to
+    // like multiple streams.  See bug 1259465.
+    test.setMediaConstraints([{audio: true}],
+                             [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoBundle.html
new file mode 100644
index 0000000000..134fa97cc0
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoBundle.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, no bundle"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.bundle = false;
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        }
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoBundleNoRtcpMux.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoBundleNoRtcpMux.html
new file mode 100644
index 0000000000..06a3a3c980
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoBundleNoRtcpMux.html
@@ -0,0 +1,44 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, no bundle and disabled RTCP-Mux"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.bundle = false;
+    options.rtcpmux = false;
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        }
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoRtcpMux.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoRtcpMux.html
new file mode 100644
index 0000000000..5d4780211a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoRtcpMux.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, with disabled RTCP-Mux"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.rtcpmux = false;
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        }
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restrictBandwidthTargetBitrate.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restrictBandwidthTargetBitrate.html
new file mode 100644
index 0000000000..ff9fb1fc22
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restrictBandwidthTargetBitrate.html
@@ -0,0 +1,29 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "Bug 1404250",
+    title: "Extremely bitrate restricted video-only peer connection"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    test.chain.insertAfter('PC_REMOTE_GET_OFFER', [
+      function PC_REMOTE_ADD_TIAS(test) {
+        test._local_offer.sdp = sdputils.addTiasBps(
+          test._local_offer.sdp, 25000);
+        info("Offer with TIAS: " + JSON.stringify(test._local_offer));
+      }
+    ]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restrictBandwidthWithTias.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restrictBandwidthWithTias.html
new file mode 100644
index 0000000000..85b831e9a8
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restrictBandwidthWithTias.html
@@ -0,0 +1,30 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1359854",
+    title: "500Kb restricted video-only peer connection"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    test.chain.insertAfter('PC_REMOTE_GET_OFFER', [
+      function PC_REMOTE_ADD_TIAS(test) {
+        test._local_offer.sdp = sdputils.addTiasBps(
+          test._local_offer.sdp, 250000);
+        info("Offer with TIAS: " + JSON.stringify(test._local_offer));
+      }
+    ]);
+    // TODO it would be nice to verify the used bandwidth
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_rtcp_rsize.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_rtcp_rsize.html
new file mode 100644
index 0000000000..25270984ea
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_rtcp_rsize.html
@@ -0,0 +1,81 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+  <script type="application/javascript" src="sdpUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1279153",
+    title: "rtcp-rsize",
+    visible: true
+  });
+
+  // 0) Use webrtc-sdp
+  // 1) ADD RTCP-RISZE to all video m-sections
+  // 2) Check for RTCP-RSIZE in ANSWER
+  // 3) Wait for media to flow
+  // 4) Wait for RTCP stats
+
+  runNetworkTest(async function (options) {
+    const test = new PeerConnectionTest(options);
+
+    let mSectionsAltered = 0;
+
+    test.chain.insertAfter("PC_LOCAL_CREATE_OFFER", [
+      function PC_LOCAL_ADD_RTCP_RSIZE(test) {
+        const lines = test.originalOffer.sdp.split("\r\n");
+        info(`SDP before rtcp-rsize: ${lines.join('\n')}`);
+        // Insert an rtcp-rsize for each m section
+        const rsizeAdded = lines.flatMap(line => {
+          if (line.startsWith("m=video")) {
+            mSectionsAltered = mSectionsAltered + 1;
+            return [line, "a=rtcp-rsize"];
+          }
+          return [line];
+        });
+        test.originalOffer.sdp = rsizeAdded.join("\r\n");
+        info(`SDP with rtcp-rsize: ${rsizeAdded.join("\n")}`);
+        is(mSectionsAltered, 1, "We only altered 1 msection")
+    }]);
+
+    // Check that the rtcp-rsize makes into the answer
+    test.chain.insertAfter("PC_LOCAL_SET_REMOTE_DESCRIPTION", [
+      function PC_LOCAL_CHECK_RTCP_RSIZE(test) {
+        const msections = sdputils.getMSections(test.pcLocal._pc.currentRemoteDescription.sdp);
+        var alteredMSectionsFound = 0;
+        for (msection of msections) {
+          if (msection.startsWith("m=video")) {
+            ok(msection.includes("\r\na=rtcp-rsize\r\n"), "video m-section includes RTCP-RSIZE");
+            alteredMSectionsFound = alteredMSectionsFound + 1;
+          } else {
+            ok(!msection.includes("\r\na=rtcp-rsize\r\n"), "audio m-section does not include RTCP-RSIZE");
+          }
+        }
+        is(alteredMSectionsFound, mSectionsAltered, "correct number of msections found");
+      }
+    ]);
+
+    // Make sure that we are still getting RTCP stats
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      async function PC_LOCAL_AND_REMOTE_CHECK_FOR_RTCP_STATS(test) {
+        await Promise.all([
+          waitForSyncedRtcp(test.pcLocal._pc),
+          waitForSyncedRtcp(test.pcRemote._pc),
+        ]);
+        // The work is done by waitForSyncedRtcp which will throw if
+        // RTCP stats are not received.
+        info("RTCP stats received!");
+      },
+    );
+    test.setMediaConstraints([{audio: true}, {video: true}], []);
+    await test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_scaleResolution.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_scaleResolution.html
new file mode 100644
index 0000000000..4be6873fa6
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_scaleResolution.html
@@ -0,0 +1,119 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1244913",
+    title: "Scale resolution down on a PeerConnection",
+    visible: true
+  });
+
+  async function checkForH264Support() {
+    const pc = new RTCPeerConnection();
+    const offer = await pc.createOffer({offerToReceiveVideo: true});
+    return offer.sdp.match(/a=rtpmap:[1-9][0-9]* H264/);
+  }
+
+  let resolutionAlignment = 1;
+
+  var mustRejectWith = (msg, reason, f) =>
+    f().then(() => ok(false, msg),
+             e => is(e.name, reason, msg));
+
+  async function testScale(codec) {
+    var pc1 = new RTCPeerConnection();
+    var pc2 = new RTCPeerConnection();
+
+    var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
+    pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+    info("testing scaling with " + codec);
+
+    let stream = await navigator.mediaDevices.getUserMedia({ video: true });
+
+    var v1 = createMediaElement('video', 'v1');
+    var v2 = createMediaElement('video', 'v2');
+
+    var ontrackfired = new Promise(resolve => pc2.ontrack = e => resolve(e));
+    var v2loadedmetadata = new Promise(resolve => v2.onloadedmetadata = resolve);
+
+    is(v2.currentTime, 0, "v2.currentTime is zero at outset");
+
+    v1.srcObject = stream;
+    var sender = pc1.addTrack(stream.getVideoTracks()[0], stream);
+    let parameters = sender.getParameters();
+    is(parameters.encodings.length, 1, "Default number of encodings should be 1");
+    parameters.encodings[0].scaleResolutionDownBy = 0.5;
+
+    await mustRejectWith(
+        "Invalid scaleResolutionDownBy must reject", "RangeError",
+        () => sender.setParameters(parameters)
+    );
+
+    parameters = sender.getParameters();
+    parameters.encodings[0].scaleResolutionDownBy = 2;
+    parameters.encodings[0].maxBitrate = 60000;
+
+    await sender.setParameters(parameters);
+
+    parameters = sender.getParameters();
+    is(parameters.encodings[0].scaleResolutionDownBy, 2, "Should be able to set scaleResolutionDownBy");
+    is(parameters.encodings[0].maxBitrate, 60000, "Should be able to set maxBitrate");
+
+    let offer = await pc1.createOffer();
+    if (codec == "VP8") {
+      offer.sdp = sdputils.removeAllButPayloadType(offer.sdp, 126);
+    }
+    await pc1.setLocalDescription(offer);
+    await pc2.setRemoteDescription(pc1.localDescription);
+
+    let answer = await pc2.createAnswer();
+    await pc2.setLocalDescription(answer);
+    await pc1.setRemoteDescription(pc2.localDescription);
+    let trackevent = await ontrackfired;
+
+    v2.srcObject = trackevent.streams[0];
+
+    await v2loadedmetadata;
+
+    await waitUntil(() => v2.currentTime > 0);
+    ok(v2.currentTime > 0, "v2.currentTime is moving (" + v2.currentTime + ")");
+
+    ok(v1.videoWidth > 0, "source width is positive");
+    ok(v1.videoHeight > 0, "source height is positive");
+    const expectedWidth =
+      v1.videoWidth / 2 - (v1.videoWidth / 2 % resolutionAlignment);
+    const expectedHeight =
+      v1.videoHeight / 2 - (v1.videoHeight / 2 % resolutionAlignment);
+    is(v2.videoWidth, expectedWidth,
+       "sink is half the width of the source");
+    is(v2.videoHeight, expectedHeight,
+       "sink is half the height of the source");
+    stream.getTracks().forEach(track => track.stop());
+    v1.srcObject = v2.srcObject = null;
+    pc1.close()
+    pc2.close()
+  }
+
+  runNetworkTest(async () => {
+    await matchPlatformH264CodecPrefs();
+    const hasH264 = await checkForH264Support();
+    if (hasH264 && navigator.userAgent.includes("Android")) {
+      // Android only has a hw encoder for h264
+      resolutionAlignment = 16;
+    }
+    await pushPrefs(['media.peerconnection.video.lock_scaling', true]);
+    await testScale("VP8");
+    if (hasH264) {
+      await testScale("H264");
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_scaleResolution_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_scaleResolution_oldSetParameters.html
new file mode 100644
index 0000000000..85a989ba32
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_scaleResolution_oldSetParameters.html
@@ -0,0 +1,122 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1244913",
+    title: "Scale resolution down on a PeerConnection",
+    visible: true
+  });
+
+  async function checkForH264Support() {
+    const pc = new RTCPeerConnection();
+    const offer = await pc.createOffer({offerToReceiveVideo: true});
+    return offer.sdp.match(/a=rtpmap:[1-9][0-9]* H264/);
+  }
+
+  let resolutionAlignment = 1;
+
+  var mustRejectWith = (msg, reason, f) =>
+    f().then(() => ok(false, msg),
+             e => is(e.name, reason, msg));
+
+  async function testScale(codec) {
+    var pc1 = new RTCPeerConnection();
+    var pc2 = new RTCPeerConnection();
+
+    var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
+    pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+    info("testing scaling with " + codec);
+
+    let stream = await navigator.mediaDevices.getUserMedia({ video: true });
+
+    var v1 = createMediaElement('video', 'v1');
+    var v2 = createMediaElement('video', 'v2');
+
+    var ontrackfired = new Promise(resolve => pc2.ontrack = e => resolve(e));
+    var v2loadedmetadata = new Promise(resolve => v2.onloadedmetadata = resolve);
+
+    is(v2.currentTime, 0, "v2.currentTime is zero at outset");
+
+    v1.srcObject = stream;
+    var sender = pc1.addTrack(stream.getVideoTracks()[0], stream);
+
+    const otherErrorStart = await GleanTest.rtcrtpsenderSetparameters.failOther.testGetValue();
+    const noTransactionIdWarningStart = await GleanTest.rtcrtpsenderSetparameters.warnNoTransactionid.testGetValue();
+
+    await mustRejectWith(
+        "Invalid scaleResolutionDownBy must reject", "RangeError",
+        () => sender.setParameters(
+            { encodings:[{ scaleResolutionDownBy: 0.5 } ] })
+    );
+
+    const otherErrorEnd = await GleanTest.rtcrtpsenderSetparameters.failOther.testGetValue();
+    const noTransactionIdWarningEnd = await GleanTest.rtcrtpsenderSetparameters.warnNoTransactionid.testGetValue();
+
+    // Make sure Glean is recording these statistics
+    is(otherErrorEnd.denominator, otherErrorStart.denominator, "No new RTCRtpSenders were created during this time");
+    is(otherErrorEnd.numerator, otherErrorStart.numerator + 1, "RTCRtpSender.setParameters reported a failure via Glean");
+    is(noTransactionIdWarningEnd.denominator, noTransactionIdWarningStart.denominator, "No new RTCRtpSenders were created during this time");
+    is(noTransactionIdWarningEnd.numerator, noTransactionIdWarningStart.numerator + 1, "Glean should have recorded a warning due to missing transactionId");
+
+    await sender.setParameters({ encodings: [{ maxBitrate: 60000,
+                                               scaleResolutionDownBy: 2 }] });
+
+    let offer = await pc1.createOffer();
+    if (codec == "VP8") {
+      offer.sdp = sdputils.removeAllButPayloadType(offer.sdp, 126);
+    }
+    await pc1.setLocalDescription(offer);
+    await pc2.setRemoteDescription(pc1.localDescription);
+
+    let answer = await pc2.createAnswer();
+    await pc2.setLocalDescription(answer);
+    await pc1.setRemoteDescription(pc2.localDescription);
+    let trackevent = await ontrackfired;
+
+    v2.srcObject = trackevent.streams[0];
+
+    await v2loadedmetadata;
+
+    await waitUntil(() => v2.currentTime > 0);
+    ok(v2.currentTime > 0, "v2.currentTime is moving (" + v2.currentTime + ")");
+
+    ok(v1.videoWidth > 0, "source width is positive");
+    ok(v1.videoHeight > 0, "source height is positive");
+    const expectedWidth =
+      v1.videoWidth / 2 - (v1.videoWidth / 2 % resolutionAlignment);
+    const expectedHeight =
+      v1.videoHeight / 2 - (v1.videoHeight / 2 % resolutionAlignment);
+    is(v2.videoWidth, expectedWidth,
+       "sink is half the width of the source");
+    is(v2.videoHeight, expectedHeight,
+       "sink is half the height of the source");
+    stream.getTracks().forEach(track => track.stop());
+    v1.srcObject = v2.srcObject = null;
+    pc1.close()
+    pc2.close()
+  }
+
+  runNetworkTest(async () => {
+    await matchPlatformH264CodecPrefs();
+    const hasH264 = await checkForH264Support();
+    if (hasH264 && navigator.userAgent.includes("Android")) {
+      // Android only has a hw encoder for h264
+      resolutionAlignment = 16;
+    }
+    await pushPrefs(['media.peerconnection.video.lock_scaling', true]);
+    await testScale("VP8");
+    if (hasH264) {
+      await testScale("H264");
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_sender_and_receiver_stats.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_sender_and_receiver_stats.html
new file mode 100644
index 0000000000..72749e8c50
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_sender_and_receiver_stats.html
@@ -0,0 +1,73 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1355220",
+    title: "RTCRtpSender.getStats() and RTCRtpReceiver.getStats()",
+    visible: true
+  });
+
+  const checkStats = (sndReport, rcvReport, kind) => {
+    ok(sndReport instanceof window.RTCStatsReport, "sender stats are a RTCStatsReport");
+    ok(rcvReport instanceof window.RTCStatsReport, "receiver stats are a RTCStatsReport");
+    // Returns SSRCs and performs some checks
+    let getSsrcs = (report, kind) => {
+      return [...report.values()]
+        .filter(stat => stat.type.endsWith("bound-rtp")).map(stat =>{
+          isnot(parseInt(stat.id, 16), NaN,
+            `id ${stat.id} is an opaque (hex) number`);
+          is(stat.kind, kind, "kind of " + stat.id
+              + " is expected type " + kind);
+          return stat.ssrc;
+      }).sort().join("|");
+    };
+    let sndSsrcs = getSsrcs(sndReport, kind);
+    let rcvSsrcs = getSsrcs(rcvReport, kind);
+    ok(sndSsrcs, "sender SSRCs is not empty");
+    ok(rcvSsrcs, "receiver SSRCs is not empty");
+    is(sndSsrcs, rcvSsrcs, "sender SSRCs match receiver SSRCs");
+  };
+
+  // This MUST be run after PC_*_WAIT_FOR_MEDIA_FLOW to ensure that we have RTP
+  // before checking for RTCP.
+  // It will throw UnsyncedRtcpError if it times out waiting for sync.
+
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      async function PC_LOCAL_AND_REMOTE_CHECK_SENDER_RECEIVER_STATS(test) {
+        await Promise.all([
+          waitForSyncedRtcp(test.pcLocal._pc),
+          waitForSyncedRtcp(test.pcRemote._pc),
+        ]);
+        let senders = test.pcLocal.getSenders();
+        let receivers = test.pcRemote.getReceivers();
+        is(senders.length, 2, "Have exactly two senders.");
+        is(receivers.length, 2, "Have exactly two receivers.");
+        for(let kind of ["audio", "video"]) {
+          let senderStats =
+              await senders.find(s => s.track.kind == kind).getStats();
+          is(senders.filter(s => s.track.kind == kind).length, 1,
+              "Exactly 1 sender of kind " + kind);
+          let receiverStats =
+              await receivers.find(r => r.track.kind == kind).getStats();
+          is(receivers.filter(r => r.track.kind == kind).length, 1,
+              "Exactly 1 receiver of kind " + kind);
+
+          checkStats(senderStats, receiverStats, kind);
+        }
+      }
+    );
+    test.setMediaConstraints([{audio: true}, {video: true}], []);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalAnswerInHaveLocalOffer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalAnswerInHaveLocalOffer.html
new file mode 100644
index 0000000000..07cdd7d6bd
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalAnswerInHaveLocalOffer.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "784519",
+    title: "setLocalDescription (answer) in 'have-local-offer'"
+  });
+
+runNetworkTest(function () {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_LOCAL_SET_LOCAL_DESCRIPTION");
+
+  test.chain.append([
+    function PC_LOCAL_SET_LOCAL_ANSWER(test) {
+      test.pcLocal._latest_offer.type = "answer";
+      return test.pcLocal.setLocalDescriptionAndFail(test.pcLocal._latest_offer)
+        .then(err => {
+          is(err.name, "InvalidModificationError", "Error is InvalidModificationError");
+        });
+    }
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalAnswerInStable.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalAnswerInStable.html
new file mode 100644
index 0000000000..e57c0640f4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalAnswerInStable.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "784519",
+    title: "setLocalDescription (answer) in 'stable'"
+  });
+
+runNetworkTest(function () {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_LOCAL_CREATE_OFFER");
+
+  test.chain.append([
+    function PC_LOCAL_SET_LOCAL_ANSWER(test) {
+      test.pcLocal._latest_offer.type = "answer";
+      return test.pcLocal.setLocalDescriptionAndFail(test.pcLocal._latest_offer)
+        .then(err => {
+          is(err.name, "InvalidModificationError", "Error is InvalidModificationError");
+        });
+    }
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalOfferInHaveRemoteOffer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalOfferInHaveRemoteOffer.html
new file mode 100644
index 0000000000..bd98a83635
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalOfferInHaveRemoteOffer.html
@@ -0,0 +1,31 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "784519",
+    title: "setLocalDescription (offer) in 'have-remote-offer'"
+  });
+
+runNetworkTest(function () {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_REMOTE_SET_REMOTE_DESCRIPTION");
+
+  test.chain.append([
+    async function PC_REMOTE_SET_LOCAL_OFFER(test) {
+      const err = await test.pcRemote.setLocalDescriptionAndFail(test.pcLocal._latest_offer);
+      is(err.name, "InvalidModificationError", "Error is InvalidModificationError");
+    }
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters.html
new file mode 100644
index 0000000000..5df97e39f5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters.html
@@ -0,0 +1,470 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1230184",
+  title: "Set parameters on sender",
+  visible: true
+});
+
+const simulcastOffer = `v=0
+o=- 3840232462471583827 0 IN IP4 127.0.0.1
+s=-
+t=0 0
+a=group:BUNDLE 0
+a=msid-semantic: WMS
+m=video 9 UDP/TLS/RTP/SAVPF 96
+c=IN IP4 0.0.0.0
+a=rtcp:9 IN IP4 0.0.0.0
+a=ice-ufrag:Li6+
+a=ice-pwd:3C05CTZBRQVmGCAq7hVasHlT
+a=ice-options:trickle
+a=fingerprint:sha-256 5B:D3:8E:66:0E:7D:D3:F3:8E:E6:80:28:19:FC:55:AD:58:5D:B9:3D:A8:DE:45:4A:E7:87:02:F8:3C:0B:3B:B3
+a=setup:actpass
+a=mid:0
+a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
+a=recvonly
+a=rtcp-mux
+a=rtpmap:96 VP8/90000
+a=rtcp-fb:96 goog-remb
+a=rtcp-fb:96 transport-cc
+a=rtcp-fb:96 ccm fir
+a=rid:foo recv
+a=rid:bar recv
+a=simulcast:recv foo;bar
+`;
+
+function buildMaximumSendEncodings() {
+  const sendEncodings = [];
+  while (true) {
+    // isDeeply does not see identical string primitives and String objects
+    // as the same, so we make this a string primitive.
+    sendEncodings.push({rid: `${sendEncodings.length}`});
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {sendEncodings});
+    const {encodings} = sender.getParameters();
+    if (encodings.length < sendEncodings.length) {
+      sendEncodings.pop();
+      return sendEncodings;
+    }
+  }
+}
+
+async function queueAWebrtcTask() {
+  const pc = new RTCPeerConnection();
+  pc.addTransceiver('audio');
+  await new Promise(r => pc.onnegotiationneeded = r);
+  pc.close();
+}
+
+// setParameters is mostly tested in wpt, but we test a few
+// implementation-specific things here. Other mochitests check whether the
+// set parameters actually have the desired effect on the media streams.
+const tests = [
+
+  // wpt currently does not assume support for 3 encodings, which limits the
+  // effectiveness of its powers-of-2 test (since it can test only for 1 and 2)
+  async function checkScaleResolutionDownByAutoFillPowersOf2() {
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    const {encodings} = sender.getParameters();
+    const scaleValues = encodings.map(({scaleResolutionDownBy}) => scaleResolutionDownBy);
+    isDeeply(scaleValues, [4, 2, 1]);
+  },
+
+  // wpt currently does not assume support for 3 encodings, which limits the
+  // effectiveness of its fill-with-1 test
+  async function checkScaleResolutionDownByAutoFillWith1() {
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [
+        {rid: "0"},{rid: "1", scaleResolutionDownBy: 3},{rid: "2"}
+      ]
+    });
+    const {encodings} = sender.getParameters();
+    const scaleValues = encodings.map(({scaleResolutionDownBy}) => scaleResolutionDownBy);
+    isDeeply(scaleValues, [1, 3, 1]);
+  },
+
+  async function checkVideoEncodingLimit() {
+    const pc = new RTCPeerConnection();
+    const maxSendEncodings = buildMaximumSendEncodings();
+    const sendEncodings = maxSendEncodings.concat({rid: "a"});
+    const {sender} = pc.addTransceiver('video', {sendEncodings});
+    const {encodings} = sender.getParameters();
+
+    const rids = encodings.map(({rid}) => rid);
+    const expectedRids = maxSendEncodings.map(({rid}) => rid);
+    isDeeply(rids, expectedRids);
+
+    const scaleValues = encodings.map(({scaleResolutionDownBy}) => scaleResolutionDownBy);
+    const expectedScaleValues = [];
+    let scale = 1;
+    while (expectedScaleValues.length < maxSendEncodings.length) {
+      expectedScaleValues.push(scale);
+      scale *= 2;
+    }
+    isDeeply(scaleValues, expectedScaleValues.reverse());
+  },
+
+  async function checkScaleDownByInTrimmedEncoding() {
+    const pc = new RTCPeerConnection();
+    const maxSendEncodings = buildMaximumSendEncodings();
+    const sendEncodings = maxSendEncodings.concat({rid: "a", scaleResolutionDownBy: 3});
+    const {sender} = pc.addTransceiver('video', {sendEncodings});
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    const expectedRids = maxSendEncodings.map(({rid}) => rid);
+    isDeeply(rids, expectedRids);
+    const scaleValues = encodings.map(({scaleResolutionDownBy}) => scaleResolutionDownBy);
+    const expectedScaleValues = maxSendEncodings.map(() => 1);
+    isDeeply(scaleValues, expectedScaleValues);
+  },
+
+  async function checkLibwebrtcRidLengthLimit() {
+    const pc = new RTCPeerConnection();
+    try {
+      pc.addTransceiver('video', {
+        sendEncodings: [{rid: "wibblywobblyjeremybearimy"}]}
+      );
+      ok(false, "Rid should be too long for libwebrtc!");
+    } catch (e) {
+      is(e.name, "TypeError",
+        "Rid that is too long for libwebrtc should result in a TypeError");
+    }
+  },
+
+  async function checkErrorsInTrimmedEncodings() {
+    const pc = new RTCPeerConnection();
+    const maxSendEncodings = buildMaximumSendEncodings();
+    try {
+      const sendEncodings = maxSendEncodings.concat({rid: "foo-bar"});
+      pc.addTransceiver('video', { sendEncodings });
+      ok(false, "Should throw due to invalid rid characters");
+    } catch (e) {
+      is(e.name, "TypeError")
+    }
+    try {
+      const sendEncodings = maxSendEncodings.concat({rid: "wibblywobblyjeremybearimy"});
+      pc.addTransceiver('video', { sendEncodings });
+      ok(false, "Should throw because rid too long");
+    } catch (e) {
+      is(e.name, "TypeError")
+    }
+    try {
+      const sendEncodings = maxSendEncodings.concat({scaleResolutionDownBy: 2});
+      pc.addTransceiver('video', { sendEncodings });
+      ok(false, "Should throw due to missing rid");
+    } catch (e) {
+      is(e.name, "TypeError")
+    }
+    try {
+      const sendEncodings = maxSendEncodings.concat(maxSendEncodings[0]);
+      pc.addTransceiver('video', { sendEncodings });
+      ok(false, "Should throw due to duplicate rid");
+    } catch (e) {
+      is(e.name, "TypeError")
+    }
+    try {
+      const sendEncodings = maxSendEncodings.concat({rid: maxSendEncodings.length, scaleResolutionDownBy: 0});
+      pc.addTransceiver('video', { sendEncodings });
+      ok(false, "Should throw due to invalid scaleResolutionDownBy");
+    } catch (e) {
+      is(e.name, "RangeError")
+    }
+    try {
+      const sendEncodings = maxSendEncodings.concat({rid: maxSendEncodings.length, maxFramerate: -1});
+      pc.addTransceiver('video', { sendEncodings });
+      ok(false, "Should throw due to invalid maxFramerate");
+    } catch (e) {
+      is(e.name, "RangeError")
+    }
+  },
+
+  async function checkCompatModeUnicastSetParametersAllowsSimulcastOffer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc1 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    const parameters = sender.getParameters();
+    parameters.encodings[0].scaleResolutionDownBy = 3.0;
+    await sender.setParameters(parameters);
+
+    await pc1.setRemoteDescription({type: "offer", sdp: simulcastOffer});
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["foo", "bar"]);
+    is(encodings[0].scaleResolutionDownBy, 2.0);
+    is(encodings[1].scaleResolutionDownBy, 1.0);
+  },
+
+  async function checkCompatModeUnicastSetParametersInterruptAllowsSimulcastOffer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc1 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    const parameters = sender.getParameters();
+    parameters.encodings[0].scaleResolutionDownBy = 3.0;
+
+    const offerDone = pc1.setRemoteDescription({type: "offer", sdp: simulcastOffer});
+    await sender.setParameters(parameters);
+    await offerDone;
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["foo", "bar"]);
+    is(encodings[0].scaleResolutionDownBy, 2.0);
+    is(encodings[1].scaleResolutionDownBy, 1.0);
+  },
+
+  async function checkCompatModeSimulcastSetParametersSetsSimulcastEnvelope() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc1 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    const parameters = sender.getParameters();
+    parameters.encodings[0].rid = "1";
+    parameters.encodings.push({rid: "2"});
+    await sender.setParameters(parameters);
+
+    await pc1.setRemoteDescription({type: "offer", sdp: simulcastOffer});
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    // No overlap in rids -> unicast
+    isDeeply(rids, [undefined]);
+  },
+
+  async function checkCompatModeSimulcastSetParametersRacesLocalUnicastOffer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    // unicast offer
+    const offer = await pc1.createOffer();
+    const aTask = queueAWebrtcTask();
+    const sldPromise =  pc1.setLocalDescription(offer);
+
+    // Right now, we have aTask queued. The success task for sLD is not queued
+    // yet, because Firefox performs the initial steps on the microtask queue,
+    // which we have not allowed to run yet. Awaiting aTask will first clear
+    // the microtask queue, then run the task queue until aTask is finished.
+    // That _should_ result in the success task for sLD(offer) being queued.
+    await aTask;
+
+    const parameters = sender.getParameters();
+    parameters.encodings[0].rid = "foo";
+    parameters.encodings.push({rid: "bar"});
+    // simulcast setParameters; the task to update [[SendEncodings]] should be
+    // queued after the success task for sLD(offer)
+    await sender.setParameters(parameters);
+    await sldPromise;
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    // Compat mode lets this slide, but won't try to negotiate it since we've
+    // already applied a unicast local offer.
+    isDeeply(rids, ["foo", "bar"]);
+
+    // Let negotiation finish, so we can generate a new offer
+    await pc2.setRemoteDescription(pc1.localDescription);
+    await pc2.setLocalDescription();
+    await pc1.setRemoteDescription(pc2.localDescription);
+
+    const reoffer = await pc1.createOffer();
+    ok(!reoffer.sdp.includes("a=simulcast"), "reoffer should be unicast");
+  },
+
+  async function checkCompatModeSimulcastSetParametersRacesRemoteOffer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc1 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    const parameters = sender.getParameters();
+    parameters.encodings[0].rid = "foo";
+    parameters.encodings.push({rid: "bar"});
+    const p = sender.setParameters(parameters);
+    await pc1.setRemoteDescription({type: "offer", sdp: simulcastOffer});
+    await p;
+    const answer = await pc1.createAnswer();
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["foo", "bar"]);
+    ok(answer.sdp.includes("a=simulcast:send foo;bar"), "answer should be simulcast");
+  },
+
+  async function checkCompatModeSimulcastSetParametersRacesLocalAnswer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    // We do an initial negotiation, and while the local answer is pending,
+    // perform a setParameters on a not-yet-negotiated video sender. The intent
+    // here is to have the success task for sLD(answer) run while the
+    // setParameters is pending.
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+
+    const audioStream = await navigator.mediaDevices.getUserMedia({audio: true});
+    // We use this later on, but set it up now so we don't inadvertently
+    // crank the event loop more than we intend below.
+    const videoStream = await navigator.mediaDevices.getUserMedia({video: true});
+    pc2.addTrack(audioStream.getTracks()[0]);
+    await pc2.setLocalDescription();
+    await pc1.setRemoteDescription(pc2.localDescription);
+    const answer = await pc1.createAnswer();
+    const aTask = queueAWebrtcTask();
+    const sldPromise = pc1.setLocalDescription(answer);
+
+    // Right now, we have aTask queued. The success task for sLD is not queued
+    // yet, because Firefox performs the initial steps on the microtask queue,
+    // which we have not allowed to run yet. Awaiting aTask will first clear
+    // the microtask queue, then run the task queue until aTask is finished.
+    // That _should_ result in the success task for sLD(answer) being queued.
+    await aTask;
+
+    // The success task for sLD(answer) should be queued now. Don't relinquish
+    // the event loop!
+
+    // New sender that has nothing to do with the negotiation in progress.
+    const sender = pc1.addTrack(videoStream.getTracks()[0]);
+    const parameters = sender.getParameters();
+    parameters.encodings[0].rid = "foo";
+    parameters.encodings.push({rid: "bar"});
+
+    // We have not relinquished the event loop, so the sLD(answer) should still
+    // be queued. The task that updates [[SendEncodings]] (from setParameters)
+    // should be queued behind it. Let them both run.
+    await sender.setParameters(parameters);
+    await sldPromise;
+
+    const offer = await pc1.createOffer();
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["foo", "bar"]);
+    ok(offer.sdp.includes("a=simulcast:send foo;bar"), "offer should be simulcast");
+  },
+
+  async function checkCompatModeSimulcastSetParametersRacesRemoteAnswer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    // We do an initial negotiation, and while the remote answer is pending,
+    // perform a setParameters on a not-yet-negotiated video sender. The intent
+    // here is to have the success task for sRD(answer) run while the
+    // setParameters is pending.
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+
+    const audioStream = await navigator.mediaDevices.getUserMedia({audio: true});
+    // We use this later on, but set it up now so we don't inadvertently
+    // crank the event loop more than we intend below.
+    const videoStream = await navigator.mediaDevices.getUserMedia({video: true});
+    pc1.addTrack(audioStream.getTracks()[0]);
+    await pc1.setLocalDescription();
+    await pc2.setRemoteDescription(pc1.localDescription);
+    await pc2.setLocalDescription();
+    const aTask = queueAWebrtcTask();
+    const srdPromise = pc1.setRemoteDescription(pc2.localDescription);
+
+    // Right now, we have aTask queued. The success task for sRD is not queued
+    // yet, because Firefox performs the initial steps on the microtask queue,
+    // which we have not allowed to run yet. Awaiting aTask will first clear
+    // the microtask queue, then run the task queue until aTask is finished.
+    // That _should_ result in the success task for sRD(answer) being queued.
+    await aTask;
+
+    // The success task for sRD(answer) should be queued now. Don't relinquish
+    // the event loop!
+
+    const sender = pc1.addTrack(videoStream.getTracks()[0]);
+    const parameters = sender.getParameters();
+    parameters.encodings[0].rid = "foo";
+    parameters.encodings.push({rid: "bar"});
+
+    // We have not relinquished the event loop, so the sRD(answer) should still
+    // be queued. The task that updates [[SendEncodings]] (from setParameters)
+    // should be queued behind it. Let them both run.
+    await sender.setParameters(parameters);
+    await srdPromise;
+
+    const offer = await pc1.createOffer();
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["foo", "bar"]);
+    ok(offer.sdp.includes("a=simulcast:send foo;bar"), "offer should be simulcast");
+  },
+
+  async function checkCompatModeSimulcastRidlessSetParametersRacesLocalOffer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    // unicast offer
+    const aTask = queueAWebrtcTask();
+    const sldPromise =  pc1.setLocalDescription();
+
+    // Right now, we have aTask queued. The success task for sLD is not queued
+    // yet, because Firefox performs the initial steps on the microtask queue,
+    // which we have not allowed to run yet. Awaiting aTask will first clear
+    // the microtask queue, then run the task queue until aTask is finished.
+    // That _should_ result in the success task for sLD(offer) being queued.
+    await aTask;
+
+    // simulcast setParameters; the task to update [[SendEncodings]] should be
+    // queued after the success task for sLD(offer)
+    try {
+      await sender.setParameters({"encodings": [{}, {}]});
+      ok(false, "setParameters with two ridless encodings should fail");
+    } catch (e) {
+      ok(true, "setParameters with two ridless encodings should fail");
+    }
+    await sldPromise;
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    // Compat mode lets this slide, but won't try to negotiate it since we've
+    // already applied a unicast local offer.
+    isDeeply(rids, [undefined]);
+
+    // Let negotiation finish, so we can generate a new offer
+    await pc2.setRemoteDescription(pc1.localDescription);
+    await pc2.setLocalDescription();
+    await pc1.setRemoteDescription(pc2.localDescription);
+
+    const reoffer = await pc1.createOffer();
+    ok(!reoffer.sdp.includes("a=simulcast"), "reoffer should be unicast");
+  },
+
+];
+
+runNetworkTest(async () => {
+  await pushPrefs(
+    ["media.peerconnection.allow_old_setParameters", false]);
+  for (const test of tests) {
+    info(`Running test: ${test.name}`);
+    await test();
+    info(`Done running test: ${test.name}`);
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_maxFramerate.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_maxFramerate.html
new file mode 100644
index 0000000000..8047719775
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_maxFramerate.html
@@ -0,0 +1,63 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1611957",
+  title: "Live-updating maxFramerate"
+});
+
+let sender, receiver;
+
+async function checkMaxFrameRate(rate) {
+  const parameters = sender.getParameters();
+  parameters.encodings[0].maxFramerate = rate;
+  await sender.setParameters(parameters);
+  await wait(2000);
+  const stats = Array.from((await receiver.getStats()).values());
+  const inboundRtp = stats.find(stat => stat.type == "inbound-rtp");
+  info(`inbound-rtp stats: ${JSON.stringify(inboundRtp)}`);
+  const fps = inboundRtp.framesPerSecond;
+  ok(fps <= (rate * 1.1) + 1,
+    `fps is an appropriate value (${fps}) for rate (${rate})`);
+}
+
+runNetworkTest(async function (options) {
+  let test = new PeerConnectionTest(options);
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.append([
+    function CHECK_PRECONDITIONS() {
+      is(test.pcLocal._pc.getSenders().length, 1,
+        "Should have 1 local sender");
+      is(test.pcRemote._pc.getReceivers().length, 1,
+        "Should have 1 remote receiver");
+
+      sender = test.pcLocal._pc.getSenders()[0];
+      receiver = test.pcRemote._pc.getReceivers()[0];
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_2() {
+      return checkMaxFrameRate(2);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_4() {
+      return checkMaxFrameRate(4);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_15() {
+      return checkMaxFrameRate(15);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_8() {
+      return checkMaxFrameRate(8);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_1() {
+      return checkMaxFrameRate(1);
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_maxFramerate_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_maxFramerate_oldSetParameters.html
new file mode 100644
index 0000000000..9c68a31c0a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_maxFramerate_oldSetParameters.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1611957",
+  title: "Live-updating maxFramerate"
+});
+
+let sender, receiver;
+
+async function checkMaxFrameRate(rate) {
+  sender.setParameters({ encodings: [{ maxFramerate: rate }] });
+  await wait(2000);
+  const stats = Array.from((await receiver.getStats()).values());
+  const inboundRtp = stats.find(stat => stat.type == "inbound-rtp");
+  info(`inbound-rtp stats: ${JSON.stringify(inboundRtp)}`);
+  const fps = inboundRtp.framesPerSecond;
+  ok(fps <= (rate * 1.1) + 1, `fps is an appropriate value (${fps}) for rate (${rate})`);
+}
+
+runNetworkTest(async function (options) {
+  let test = new PeerConnectionTest(options);
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.append([
+    function CHECK_PRECONDITIONS() {
+      is(test.pcLocal._pc.getSenders().length, 1,
+          "Should have 1 local sender");
+      is(test.pcRemote._pc.getReceivers().length, 1,
+          "Should have 1 remote receiver");
+
+      sender = test.pcLocal._pc.getSenders()[0];
+      receiver = test.pcRemote._pc.getReceivers()[0];
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_2() {
+      return checkMaxFrameRate(2);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_4() {
+      return checkMaxFrameRate(4);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_15() {
+      return checkMaxFrameRate(15);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_8() {
+      return checkMaxFrameRate(8);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_1() {
+      return checkMaxFrameRate(1);
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_oldSetParameters.html
new file mode 100644
index 0000000000..2b55ec46e6
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_oldSetParameters.html
@@ -0,0 +1,86 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1230184",
+  title: "Set parameters on sender",
+  visible: true
+});
+
+function parameterstest(pc) {
+  ok(pc.getSenders().length, "have senders");
+  var sender = pc.getSenders()[0];
+
+  var testParameters = (params, errorName, errorMsg) => {
+    info("Trying to set " + JSON.stringify(params));
+
+    var validateParameters = (a, b) => {
+      var validateEncoding = (a, b) => {
+        is(a.rid, b.rid, "same rid");
+        is(a.maxBitrate, b.maxBitrate, "same maxBitrate");
+        is(a.maxFramerate, b.maxFramerate, "same maxFramerate");
+        is(a.scaleResolutionDownBy, b.scaleResolutionDownBy,
+           "same scaleResolutionDownBy");
+      };
+      is(a.encodings.length, (b.encodings || []).length, "same encodings");
+      a.encodings.forEach((en, i) => validateEncoding(en, b.encodings[i]));
+    };
+
+    var before = JSON.stringify(sender.getParameters());
+    isnot(JSON.stringify(params), before, "starting condition");
+
+    var p = sender.setParameters(params)
+    .then(() => {
+      isnot(JSON.stringify(sender.getParameters()), before, "parameters changed");
+      validateParameters(sender.getParameters(), params);
+      is(null, errorName || null, "is success expected");
+    }, e => {
+      is(e.name, errorName, "correct error name");
+      is(e.message, errorMsg, "correct error message");
+    });
+    is(JSON.stringify(sender.getParameters()), before, "parameters not set yet");
+    return p;
+  };
+
+  return [
+    [{ encodings: [ { rid: "foo", maxBitrate: 40000, scaleResolutionDownBy: 2 },
+                    { rid: "bar", maxBitrate: 10000, scaleResolutionDownBy: 4 }]
+    }],
+    [{ encodings: [{ maxBitrate: 10000, scaleResolutionDownBy: 4 }]}],
+    [{ encodings: [{ maxFramerate: 0.0, scaleResolutionDownBy: 1 }]}],
+    [{ encodings: [{ maxFramerate: 30.5, scaleResolutionDownBy: 1 }]}],
+    [{ encodings: [{ maxFramerate: -1, scaleResolutionDownBy: 1 }]}, "RangeError", "maxFramerate must be non-negative"],
+    [{ encodings: [{ maxBitrate: 40000 },
+                   { rid: "bar", maxBitrate: 10000 }] }, "TypeError", "Missing rid"],
+    [{ encodings: [{ rid: "foo", maxBitrate: 40000 },
+                   { rid: "bar", maxBitrate: 10000 },
+                   { rid: "bar", maxBitrate: 20000 }] }, "TypeError", "Duplicate rid"],
+    [{}, "TypeError", `RTCRtpSender.setParameters: Missing required 'encodings' member of RTCRtpSendParameters.`]
+  ].reduce((p, args) => p.then(() => testParameters.apply(this, args)),
+           Promise.resolve());
+}
+
+runNetworkTest(() => {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{video: true}], [{video: true}]);
+  test.chain.removeAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW");
+
+  // Test sender parameters.
+  test.chain.append([
+    function PC_LOCAL_SET_PARAMETERS(test) {
+      return parameterstest(test.pcLocal._pc);
+    }
+  ]);
+
+  return test.run();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_scaleResolutionDownBy.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_scaleResolutionDownBy.html
new file mode 100644
index 0000000000..d1275d6523
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_scaleResolutionDownBy.html
@@ -0,0 +1,98 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1253499",
+  title: "Live-updating scaleResolutionDownBy"
+});
+
+async function checkForH264Support() {
+  const pc = new RTCPeerConnection();
+  const offer = await pc.createOffer({offerToReceiveVideo: true});
+  return offer.sdp.match(/a=rtpmap:[1-9][0-9]* H264/);
+}
+
+let sender, localElem, remoteElem;
+let originalWidth, originalHeight;
+let resolutionAlignment = 1;
+
+async function checkScaleDownBy(scale) {
+  const parameters = sender.getParameters();
+  parameters.encodings[0].scaleResolutionDownBy = scale;
+  await sender.setParameters(parameters);
+  await haveEvent(remoteElem, "resize", wait(5000, new Error("Timeout")));
+
+  // Find the expected resolution. Internally we floor the exact scaling, then
+  // shrink each dimension to the alignment requested by the encoder.
+  let expectedWidth =
+    originalWidth / scale - (originalWidth / scale % resolutionAlignment);
+  let expectedHeight =
+    originalHeight / scale - (originalHeight / scale % resolutionAlignment);
+
+  is(remoteElem.videoWidth, expectedWidth,
+    `Width should have scaled down by ${scale}`);
+  is(remoteElem.videoHeight, expectedHeight,
+    `Height should have scaled down by ${scale}`);
+}
+
+runNetworkTest(async function (options) {
+  await pushPrefs(['media.peerconnection.video.lock_scaling', true]);
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    if (await checkForH264Support()) {
+      // Android only has h264 in hw, so now we know it will use vp8 in hw too.
+      resolutionAlignment = 16;
+    }
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  let test = new PeerConnectionTest(options);
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.append([
+    function CHECK_PRECONDITIONS() {
+      is(test.pcLocal._pc.getSenders().length, 1,
+          "Should have 1 local sender");
+      is(test.pcLocal.localMediaElements.length, 1,
+          "Should have 1 local sending media element");
+      is(test.pcRemote.remoteMediaElements.length, 1,
+          "Should have 1 remote media element");
+
+      sender = test.pcLocal._pc.getSenders()[0];
+      localElem = test.pcLocal.localMediaElements[0];
+      remoteElem = test.pcRemote.remoteMediaElements[0];
+
+      remoteElem.addEventListener("resize", () =>
+        info(`Video resized to ${remoteElem.videoWidth}x${remoteElem.videoHeight}`));
+
+      originalWidth = localElem.videoWidth;
+      originalHeight = localElem.videoHeight;
+      info(`Original width is ${originalWidth}`);
+    },
+    function PC_LOCAL_SCALEDOWNBY_2() {
+      return checkScaleDownBy(2);
+    },
+    function PC_LOCAL_SCALEDOWNBY_4() {
+      return checkScaleDownBy(4);
+    },
+    function PC_LOCAL_SCALEDOWNBY_15() {
+      return checkScaleDownBy(15);
+    },
+    function PC_LOCAL_SCALEDOWNBY_8() {
+      return checkScaleDownBy(8);
+    },
+    function PC_LOCAL_SCALEDOWNBY_1() {
+      return checkScaleDownBy(1);
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_scaleResolutionDownBy_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_scaleResolutionDownBy_oldSetParameters.html
new file mode 100644
index 0000000000..4d515bd5c1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_scaleResolutionDownBy_oldSetParameters.html
@@ -0,0 +1,96 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1253499",
+  title: "Live-updating scaleResolutionDownBy"
+});
+
+async function checkForH264Support() {
+  const pc = new RTCPeerConnection();
+  const offer = await pc.createOffer({offerToReceiveVideo: true});
+  return offer.sdp.match(/a=rtpmap:[1-9][0-9]* H264/);
+}
+
+let sender, localElem, remoteElem;
+let originalWidth, originalHeight;
+let resolutionAlignment = 1;
+
+async function checkScaleDownBy(scale) {
+  sender.setParameters({ encodings: [{ scaleResolutionDownBy: scale }] });
+  await haveEvent(remoteElem, "resize", wait(5000, new Error("Timeout")));
+
+  // Find the expected resolution. Internally we floor the exact scaling, then
+  // shrink each dimension to the alignment requested by the encoder.
+  let expectedWidth =
+    originalWidth / scale - (originalWidth / scale % resolutionAlignment);
+  let expectedHeight =
+    originalHeight / scale - (originalHeight / scale % resolutionAlignment);
+
+  is(remoteElem.videoWidth, expectedWidth,
+    `Width should have scaled down by ${scale}`);
+  is(remoteElem.videoHeight, expectedHeight,
+    `Height should have scaled down by ${scale}`);
+}
+
+runNetworkTest(async function (options) {
+  await pushPrefs(['media.peerconnection.video.lock_scaling', true]);
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    if (await checkForH264Support()) {
+      // Android only has h264 in hw, so now we know it will use vp8 in hw too.
+      resolutionAlignment = 16;
+    }
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  let test = new PeerConnectionTest(options);
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.append([
+    function CHECK_PRECONDITIONS() {
+      is(test.pcLocal._pc.getSenders().length, 1,
+          "Should have 1 local sender");
+      is(test.pcLocal.localMediaElements.length, 1,
+          "Should have 1 local sending media element");
+      is(test.pcRemote.remoteMediaElements.length, 1,
+          "Should have 1 remote media element");
+
+      sender = test.pcLocal._pc.getSenders()[0];
+      localElem = test.pcLocal.localMediaElements[0];
+      remoteElem = test.pcRemote.remoteMediaElements[0];
+
+      remoteElem.addEventListener("resize", () =>
+        info(`Video resized to ${remoteElem.videoWidth}x${remoteElem.videoHeight}`));
+
+      originalWidth = localElem.videoWidth;
+      originalHeight = localElem.videoHeight;
+      info(`Original width is ${originalWidth}`);
+    },
+    function PC_LOCAL_SCALEDOWNBY_2() {
+      return checkScaleDownBy(2);
+    },
+    function PC_LOCAL_SCALEDOWNBY_4() {
+      return checkScaleDownBy(4);
+    },
+    function PC_LOCAL_SCALEDOWNBY_15() {
+      return checkScaleDownBy(15);
+    },
+    function PC_LOCAL_SCALEDOWNBY_8() {
+      return checkScaleDownBy(8);
+    },
+    function PC_LOCAL_SCALEDOWNBY_1() {
+      return checkScaleDownBy(1);
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteAnswerInHaveRemoteOffer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteAnswerInHaveRemoteOffer.html
new file mode 100644
index 0000000000..1912835160
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteAnswerInHaveRemoteOffer.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "784519",
+    title: "setRemoteDescription (answer) in 'have-remote-offer'"
+  });
+
+runNetworkTest(function () {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_REMOTE_SET_REMOTE_DESCRIPTION");
+
+  test.chain.append([
+    function PC_REMOTE_SET_REMOTE_ANSWER(test) {
+      test.pcLocal._latest_offer.type = "answer";
+      test.pcRemote._pc.setRemoteDescription(test.pcLocal._latest_offer)
+        .then(generateErrorCallback('setRemoteDescription should fail'),
+              err =>
+              is(err.name, "InvalidStateError", "Error is InvalidStateError"));
+    }
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteAnswerInStable.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteAnswerInStable.html
new file mode 100644
index 0000000000..6208fdea3e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteAnswerInStable.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "784519",
+    title: "setRemoteDescription (answer) in 'stable'"
+  });
+
+runNetworkTest(function () {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_LOCAL_CREATE_OFFER");
+
+  test.chain.append([
+    function PC_LOCAL_SET_REMOTE_ANSWER(test) {
+      test.pcLocal._latest_offer.type = "answer";
+      test.pcLocal._pc.setRemoteDescription(test.pcLocal._latest_offer)
+        .then(generateErrorCallback('setRemoteDescription should fail'),
+              err =>
+              is(err.name, "InvalidStateError", "Error is InvalidStateError"));
+    }
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteOfferInHaveLocalOffer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteOfferInHaveLocalOffer.html
new file mode 100644
index 0000000000..20236f442c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteOfferInHaveLocalOffer.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "784519",
+    title: "setRemoteDescription (offer) in 'have-local-offer'"
+  });
+
+runNetworkTest(function () {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_LOCAL_SET_LOCAL_DESCRIPTION");
+
+  test.chain.append([
+    async function PC_LOCAL_SET_REMOTE_OFFER(test) {
+      const p = test.pcLocal._pc.setRemoteDescription(test.pcLocal._latest_offer);
+      await new Promise(r => test.pcLocal.onsignalingstatechange = r);
+      is(test.pcLocal._pc.signalingState, 'stable', 'should fire stable');
+      await new Promise(r => test.pcLocal.onsignalingstatechange = r);
+      is(test.pcLocal._pc.signalingState, 'have-remote-offer',
+         'should fire have-remote-offer');
+      await p;
+      ok(true, 'setRemoteDescription should succeed');
+    }
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer.html
new file mode 100644
index 0000000000..ba75c72022
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer.html
@@ -0,0 +1,121 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast answer",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    offerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // Two recv transceivers, one for each simulcast stream
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = midToRid(offer);
+    info(`Transformed recv offer to simulcast: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: 'offer', sdp: mungedOffer});
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    const sender = answerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    let parameters = sender.getParameters();
+    is(parameters.encodings.length, 2);
+    is(answerer.getSenders().length, 1);
+    emitter.start();
+
+    await offerer.setLocalDescription(offer);
+
+    const rids = offerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    parameters = sender.getParameters();
+    info(`parameters: ${JSON.stringify(parameters)}`);
+    const observedRids = parameters.encodings.map(({rid}) => rid);
+    isDeeply(observedRids, rids);
+    parameters.encodings[0].maxBitrate = 40000;
+    parameters.encodings[0].scaleResolutionDownBy = 1;
+    parameters.encodings[1].maxBitrate = 40000;
+    parameters.encodings[1].scaleResolutionDownBy = 2;
+    await sender.setParameters(parameters);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = ridToMid(answer);
+    info(`Transformed send simulcast answer to multiple m-sections: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: 'answer', sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[0].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[1].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    info("Stats ready");
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_lowResFirst.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_lowResFirst.html
new file mode 100644
index 0000000000..00c6e4ad3a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_lowResFirst.html
@@ -0,0 +1,113 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast answer, first rid has lowest resolution",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    offerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // Two recv transceivers, one for each simulcast stream
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    answerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    emitter.start();
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = midToRid(offer);
+    info(`Transformed recv offer to simulcast: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: 'offer', sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = offerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const sender = answerer.getSenders()[0];
+    const parameters = sender.getParameters();
+    parameters.encodings[0].maxBitrate = 40000;
+    parameters.encodings[0].scaleResolutionDownBy = 2;
+    parameters.encodings[1].maxBitrate = 40000;
+    await sender.setParameters(parameters);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = ridToMid(answer);
+    info(`Transformed send simulcast answer to multiple m-sections: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: 'answer', sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[1].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[0].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_lowResFirst_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_lowResFirst_oldSetParameters.html
new file mode 100644
index 0000000000..c2aafc4575
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_lowResFirst_oldSetParameters.html
@@ -0,0 +1,115 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast answer, first rid has lowest resolution",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        ['media.peerconnection.simulcast', true],
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    offerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // Two recv transceivers, one for each simulcast stream
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    answerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    emitter.start();
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = midToRid(offer);
+    info(`Transformed recv offer to simulcast: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: "offer", sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = offerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const sender = answerer.getSenders()[0];
+    sender.setParameters({
+        encodings: [
+            { rid: rids[0], maxBitrate: 40000, scaleResolutionDownBy: 2 },
+            { rid: rids[1], maxBitrate: 40000 }
+          ]
+    });
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = ridToMid(answer);
+    info(`Transformed send simulcast answer to multiple m-sections: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: "answer", sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[1].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[0].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_oldSetParameters.html
new file mode 100644
index 0000000000..bc0b9f71cc
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_oldSetParameters.html
@@ -0,0 +1,115 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast answer",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        ['media.peerconnection.simulcast', true],
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    offerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // Two recv transceivers, one for each simulcast stream
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    answerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    emitter.start();
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = midToRid(offer);
+    info(`Transformed recv offer to simulcast: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: "offer", sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = offerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const sender = answerer.getSenders()[0];
+    await sender.setParameters({
+        encodings: [
+            { rid: rids[0], maxBitrate: 40000 },
+            { rid: rids[1], maxBitrate: 40000, scaleResolutionDownBy: 2 }
+          ]
+    });
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = ridToMid(answer);
+    info(`Transformed send simulcast answer to multiple m-sections: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: "answer", sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[0].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[1].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOddResolution.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOddResolution.html
new file mode 100644
index 0000000000..c380b34f1a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOddResolution.html
@@ -0,0 +1,183 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1432793",
+    title: "Simulcast with odd resolution",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    const helper = new VideoStreamHelper();
+    const emitter = new VideoFrameEmitter(helper.green, helper.red, 705, 528);
+
+    async function checkVideoElement(senderElement, receiverElement, encoding) {
+      info(`Waiting for receiver video element ${encoding.rid} to start playing`);
+      await helper.checkVideoPlaying(receiverElement);
+      const srcWidth = senderElement.videoWidth;
+      const srcHeight = senderElement.videoHeight;
+      info(`Source resolution is ${srcWidth}x${srcHeight}`);
+
+      const scaleDownBy = encoding.scaleResolutionDownBy;
+      const expectedWidth = srcWidth / scaleDownBy;
+      const expectedHeight = srcHeight / scaleDownBy;
+      const margin = srcWidth * 0.1;
+      const width = receiverElement.videoWidth;
+      const height = receiverElement.videoHeight;
+      const rid = encoding.rid;
+      ok(width >= expectedWidth - margin && width <= expectedWidth + margin,
+        `Width ${width} should be within 10% of ${expectedWidth} for rid '${rid}'`);
+      ok(height >= expectedHeight - margin && height <= expectedHeight + margin,
+        `Height ${height} should be within 10% of ${expectedHeight} for rid '${rid}'`);
+    }
+
+    async function checkVideoElements(senderElement, receiverElements, encodings) {
+      is(receiverElements.length, encodings.length, 'Number of video elements should match number of encodings');
+      info('Waiting for sender video element to start playing');
+      await helper.checkVideoPlaying(senderElement);
+      for (let i = 0; i < encodings.length; i++) {
+        await checkVideoElement(senderElement, receiverElements[i], encodings[i]);
+      }
+    }
+
+    const sendEncodings = [{ rid: "0", maxBitrate: 40000, scaleResolutionDownBy: 1.9 },
+                           { rid: "1", maxBitrate: 40000, scaleResolutionDownBy: 3.5 },
+                           { rid: "2", maxBitrate: 40000, scaleResolutionDownBy: 17.8 }];
+
+    async function checkSenderStats(sender) {
+      const senderStats = await sender.getStats();
+      checkSenderStats(senderStats, sendEncodings.length);
+      checkExpectedFields(senderStats);
+      pedanticChecks(senderStats);
+    }
+
+    async function waitForResizeEvents(elements) {
+      return Promise.all(elements.map(elem => haveEvent(elem, 'resize')));
+    }
+
+    await pushPrefs(
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const videoStream = emitter.stream();
+    offerer.addTransceiver(videoStream.getVideoTracks()[0], {sendEncodings});
+    const senderElement = document.createElement('video');
+    senderElement.autoplay = true;
+    senderElement.srcObject = videoStream;
+    senderElement.id = videoStream.id
+
+    const sender = offerer.getSenders()[0];
+    let parameters = sender.getParameters();
+    is(parameters.encodings[0].maxBitrate, sendEncodings[0].maxBitrate);
+    isfuzzy(parameters.encodings[0].scaleResolutionDownBy,
+      sendEncodings[0].scaleResolutionDownBy, 0.01);
+    is(parameters.encodings[1].maxBitrate, sendEncodings[1].maxBitrate);
+    isfuzzy(parameters.encodings[1].scaleResolutionDownBy,
+      sendEncodings[1].scaleResolutionDownBy, 0.01);
+    is(parameters.encodings[2].maxBitrate, sendEncodings[2].maxBitrate);
+    isfuzzy(parameters.encodings[2].scaleResolutionDownBy,
+      sendEncodings[2].scaleResolutionDownBy, 0.01);
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = ridToMid(offer);
+    info(`Transformed send simulcast offer to multiple m-sections: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: 'offer', sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = answerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 3, 'Should have 3 mids in offer');
+    ok(rids[0], 'First mid should be non-empty');
+    ok(rids[1], 'Second mid should be non-empty');
+    ok(rids[2], 'Third mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = midToRid(answer);
+    info(`Transformed recv answer to simulcast: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: 'answer', sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 3, 'Offerer should have gotten 3 ontrack events');
+    emitter.start();
+    info('Waiting for 3 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+    await checkVideoElements(senderElement, videoElems, parameters.encodings);
+    emitter.stop();
+
+    await Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    info(`Changing source resolution to 1280x720`);
+    emitter.size(1280, 720);
+    emitter.start();
+    await waitForResizeEvents([senderElement, ...videoElems]);
+    await checkVideoElements(senderElement, videoElems, parameters.encodings);
+    await checkSenderStats(sender);
+
+    parameters = sender.getParameters();
+    parameters.encodings[0].scaleResolutionDownBy = 1;
+    parameters.encodings[1].scaleResolutionDownBy = 2;
+    parameters.encodings[2].scaleResolutionDownBy = 3;
+    info(`Changing encodings to ${JSON.stringify(parameters.encodings)}`);
+    await sender.setParameters(parameters);
+    await waitForResizeEvents(videoElems);
+    await checkVideoElements(senderElement, videoElems, parameters.encodings);
+    await checkSenderStats(sender);
+
+    parameters = sender.getParameters();
+    parameters.encodings[0].scaleResolutionDownBy = 6;
+    parameters.encodings[1].scaleResolutionDownBy = 5;
+    parameters.encodings[2].scaleResolutionDownBy = 4;
+    info(`Changing encodings to ${JSON.stringify(parameters.encodings)}`);
+    await sender.setParameters(parameters);
+    await waitForResizeEvents(videoElems);
+    await checkVideoElements(senderElement, videoElems, parameters.encodings);
+    await checkSenderStats(sender);
+
+    parameters = sender.getParameters();
+    parameters.encodings[0].scaleResolutionDownBy = 4;
+    parameters.encodings[1].scaleResolutionDownBy = 1;
+    parameters.encodings[2].scaleResolutionDownBy = 2;
+    info(`Changing encodings to ${JSON.stringify(parameters.encodings)}`);
+    await sender.setParameters(parameters);
+    await waitForResizeEvents(videoElems);
+    await checkVideoElements(senderElement, videoElems, parameters.encodings);
+    await checkSenderStats(sender);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOddResolution_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOddResolution_oldSetParameters.html
new file mode 100644
index 0000000000..0f6d3c8520
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOddResolution_oldSetParameters.html
@@ -0,0 +1,172 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1432793",
+    title: "Simulcast with odd resolution",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    const helper = new VideoStreamHelper();
+    const emitter = new VideoFrameEmitter(helper.green, helper.red, 705, 528);
+
+    async function checkVideoElement(senderElement, receiverElement, encoding) {
+      info(`Waiting for receiver video element ${encoding.rid} to start playing`);
+      await helper.checkVideoPlaying(receiverElement);
+      const srcWidth = senderElement.videoWidth;
+      const srcHeight = senderElement.videoHeight;
+      info(`Source resolution is ${srcWidth}x${srcHeight}`);
+
+      const scaleDownBy = encoding.scaleResolutionDownBy;
+      const expectedWidth = srcWidth / scaleDownBy;
+      const expectedHeight = srcHeight / scaleDownBy;
+      const margin = srcWidth * 0.1;
+      const width = receiverElement.videoWidth;
+      const height = receiverElement.videoHeight;
+      const rid = encoding.rid;
+      ok(width >= expectedWidth - margin && width <= expectedWidth + margin,
+        `Width ${width} should be within 10% of ${expectedWidth} for rid '${rid}'`);
+      ok(height >= expectedHeight - margin && height <= expectedHeight + margin,
+        `Height ${height} should be within 10% of ${expectedHeight} for rid '${rid}'`);
+    }
+
+    async function checkVideoElements(senderElement, receiverElements, encodings) {
+      is(receiverElements.length, encodings.length, 'Number of video elements should match number of encodings');
+      info('Waiting for sender video element to start playing');
+      await helper.checkVideoPlaying(senderElement);
+      for (let i = 0; i < encodings.length; i++) {
+        await checkVideoElement(senderElement, receiverElements[i], encodings[i]);
+      }
+    }
+
+    async function checkSenderStats(sender) {
+      const senderStats = await sender.getStats();
+      checkSenderStats(senderStats, encodings.length);
+      checkExpectedFields(senderStats);
+      pedanticChecks(senderStats);
+    }
+
+    async function waitForResizeEvents(elements) {
+      return Promise.all(elements.map(elem => haveEvent(elem, 'resize')));
+    }
+
+    const encodings = [{ rid: "0", maxBitrate: 40000, scaleResolutionDownBy: 1.9 },
+                       { rid: "1", maxBitrate: 40000, scaleResolutionDownBy: 3.5 },
+                       { rid: "2", maxBitrate: 40000, scaleResolutionDownBy: 17.8 }];
+
+    await pushPrefs(
+        ['media.peerconnection.simulcast', true],
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const videoStream = emitter.stream();
+    offerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    const senderElement = document.createElement('video');
+    senderElement.autoplay = true;
+    senderElement.srcObject = videoStream;
+    senderElement.id = videoStream.id
+
+    const sender = offerer.getSenders()[0];
+    sender.setParameters({encodings});
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = ridToMid(offer);
+    info(`Transformed send simulcast offer to multiple m-sections: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: "offer", sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = answerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 3, 'Should have 3 mids in offer');
+    ok(rids[0], 'First mid should be non-empty');
+    ok(rids[1], 'Second mid should be non-empty');
+    ok(rids[2], 'Third mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = midToRid(answer);
+    info(`Transformed recv answer to simulcast: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: "answer", sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 3, 'Offerer should have gotten 3 ontrack events');
+    emitter.start();
+    info('Waiting for 3 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+    await checkVideoElements(senderElement, videoElems, encodings);
+    emitter.stop();
+
+    await Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    info(`Changing source resolution to 1280x720`);
+    emitter.size(1280, 720);
+    emitter.start();
+    await waitForResizeEvents([senderElement, ...videoElems]);
+    await checkVideoElements(senderElement, videoElems, encodings);
+    await checkSenderStats(sender);
+
+    encodings[0].scaleResolutionDownBy = 1;
+    encodings[1].scaleResolutionDownBy = 2;
+    encodings[2].scaleResolutionDownBy = 3;
+    info(`Changing encodings to ${JSON.stringify(encodings)}`);
+    await sender.setParameters({encodings});
+    await waitForResizeEvents(videoElems);
+    await checkVideoElements(senderElement, videoElems, encodings);
+    await checkSenderStats(sender);
+
+    encodings[0].scaleResolutionDownBy = 6;
+    encodings[1].scaleResolutionDownBy = 5;
+    encodings[2].scaleResolutionDownBy = 4;
+    info(`Changing encodings to ${JSON.stringify(encodings)}`);
+    await sender.setParameters({encodings});
+    await waitForResizeEvents(videoElems);
+    await checkVideoElements(senderElement, videoElems, encodings);
+    await checkSenderStats(sender);
+
+    encodings[0].scaleResolutionDownBy = 4;
+    encodings[1].scaleResolutionDownBy = 1;
+    encodings[2].scaleResolutionDownBy = 2;
+    info(`Changing encodings to ${JSON.stringify(encodings)}`);
+    await sender.setParameters({encodings});
+    await waitForResizeEvents(videoElems);
+    await checkVideoElements(senderElement, videoElems, encodings);
+    await checkSenderStats(sender);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer.html
new file mode 100644
index 0000000000..cb7c13a0d1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer.html
@@ -0,0 +1,109 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="parser_rtp.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast offer",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    const sendEncodings = [
+            { rid: '0', maxBitrate: 40000 },
+            { rid: '1', maxBitrate: 40000, scaleResolutionDownBy: 2 }
+          ];
+    offerer.addTransceiver(videoStream.getVideoTracks()[0], {sendEncodings});
+    emitter.start();
+
+    const sender = offerer.getSenders()[0];
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = ridToMid(offer);
+    info(`Transformed send simulcast offer to multiple m-sections: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: 'offer', sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = answerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = midToRid(answer);
+    info(`Transformed recv answer to simulcast: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: 'answer', sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[0].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[1].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_lowResFirst.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_lowResFirst.html
new file mode 100644
index 0000000000..93141311f1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_lowResFirst.html
@@ -0,0 +1,109 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="parser_rtp.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast offer, first rid has lowest resolution",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    const sendEncodings = [
+      { rid: '0', maxBitrate: 40000, scaleResolutionDownBy: 2 },
+      { rid: '1', maxBitrate: 40000 }
+    ];
+    offerer.addTransceiver(videoStream.getVideoTracks()[0], {sendEncodings});
+    emitter.start();
+
+    const sender = offerer.getSenders()[0];
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = ridToMid(offer);
+    info(`Transformed send simulcast offer to multiple m-sections: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: 'offer', sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = answerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = midToRid(answer);
+    info(`Transformed recv answer to simulcast: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: 'answer', sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[1].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[0].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_lowResFirst_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_lowResFirst_oldSetParameters.html
new file mode 100644
index 0000000000..73e2d38eb2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_lowResFirst_oldSetParameters.html
@@ -0,0 +1,112 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="parser_rtp.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast offer, first rid has lowest resolution",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        ['media.peerconnection.simulcast', true],
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    offerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    emitter.start();
+
+    const sender = offerer.getSenders()[0];
+    sender.setParameters({
+        encodings: [
+            { rid: '0', maxBitrate: 40000, scaleResolutionDownBy: 2 },
+            { rid: '1', maxBitrate: 40000 }
+          ]
+    });
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = ridToMid(offer);
+    info(`Transformed send simulcast offer to multiple m-sections: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: 'offer', sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = answerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = midToRid(answer);
+    info(`Transformed recv answer to simulcast: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: 'answer', sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[1].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[0].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_oldSetParameters.html
new file mode 100644
index 0000000000..551273af5e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_oldSetParameters.html
@@ -0,0 +1,112 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="parser_rtp.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast offer",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        ['media.peerconnection.simulcast', true],
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    offerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    emitter.start();
+
+    const sender = offerer.getSenders()[0];
+    sender.setParameters({
+        encodings: [
+            { rid: '0', maxBitrate: 40000 },
+            { rid: '1', maxBitrate: 40000, scaleResolutionDownBy: 2 }
+          ]
+    });
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = ridToMid(offer);
+    info(`Transformed send simulcast offer to multiple m-sections: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: "offer", sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = answerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = midToRid(answer);
+    info(`Transformed recv answer to simulcast: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: "answer", sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[0].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[1].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_stats.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats.html
new file mode 100644
index 0000000000..2ef98dc9c8
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats.html
@@ -0,0 +1,42 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1337525",
+  title: "webRtc Stats composition and sanity"
+});
+
+runNetworkTest(async function (options) {
+  // We don't know how to get QP value when using Android system codecs.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  // For accurate comparisons of `remoteTimestamp` (not using reduced precision)
+  // to `timestamp` (using reduced precision).
+  await pushPrefs(["privacy.resistFingerprinting.reduceTimerPrecision.jitter",
+                   false]);
+
+  const test = new PeerConnectionTest(options);
+
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+    [PC_LOCAL_TEST_LOCAL_STATS]);
+
+  test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+    [PC_REMOTE_TEST_REMOTE_STATS]);
+
+  test.setMediaConstraints([{audio: true}, {video: true}],
+                           [{audio: true}, {video: true}]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_jitter.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_jitter.html
new file mode 100644
index 0000000000..6e1ef698b4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_jitter.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1672590",
+    title: "Jitter sanity check"
+  });
+
+const checkJitter = stats => {
+  stats.forEach((stat, mapKey) => {
+    if (stat.type == "remote-inbound-rtp") {
+      // This should be much lower for audio, TODO: Bug 1330575
+      const expectedJitter = stat.kind == "video" ? 0.5 : 1;
+
+      ok(stat.jitter < expectedJitter,
+          stat.type + ".jitter is sane number for a local only test. value="
+          + stat.jitter);
+    }
+  });
+};
+
+const PC_LOCAL_TEST_LOCAL_JITTER = async test => {
+  checkJitter(await waitForSyncedRtcp(test.pcLocal._pc));
+}
+
+const PC_REMOTE_TEST_REMOTE_JITTER = async test => {
+  checkJitter(await waitForSyncedRtcp(test.pcRemote._pc));
+}
+
+runNetworkTest(async function (options) {
+  // We don't know how to get QP value when using Android system codecs.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  const test = new PeerConnectionTest(options);
+
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+    [PC_LOCAL_TEST_LOCAL_JITTER]);
+
+  test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+    [PC_REMOTE_TEST_REMOTE_JITTER]);
+
+  test.setMediaConstraints([{audio: true}, {video: true}],
+                           [{audio: true}, {video: true}]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_oneway.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_oneway.html
new file mode 100644
index 0000000000..02ace530a9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_oneway.html
@@ -0,0 +1,65 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1225722",
+  title: "WebRTC Stats composition and sanity for a one-way peer connection"
+});
+
+runNetworkTest(async function (options) {
+  // We don't know how to get QP value when using Android system codecs.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  // For accurate comparisons of `remoteTimestamp` (not using reduced precision)
+  // to `timestamp` (using reduced precision).
+  await pushPrefs(["privacy.resistFingerprinting.reduceTimerPrecision.jitter",
+                   false]);
+
+  const test = new PeerConnectionTest(options);
+
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+    [PC_LOCAL_TEST_LOCAL_STATS]);
+
+  test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+    [PC_REMOTE_TEST_REMOTE_STATS]);
+
+  const testOneWayStats = (stats, codecType) => {
+    const codecs = [];
+    stats.forEach(stat => {
+      if (stat.type == "codec") {
+        codecs.push(stat);
+        is(stat.codecType, codecType, "One-way codec has specific codecType");
+      }
+    });
+    is(codecs.length, 2, "One audio and one video codec");
+    if (codecs.length == 2) {
+      isnot(codecs[0].mimeType.slice(0, 5), codecs[1].mimeType.slice(0, 5),
+        "Different media type for audio vs video mime types");
+    }
+  };
+
+  test.chain.append([
+    async function PC_LOCAL_TEST_CODECTYPE_ENCODE(test) {
+      testOneWayStats(await test.pcLocal._pc.getStats(), "encode");
+    },
+    async function PC_REMOTE_TEST_CODECTYPE_DECODE(test) {
+      testOneWayStats(await test.pcRemote._pc.getStats(), "decode");
+    },
+  ]);
+
+  test.setMediaConstraints([{audio: true}, {video: true}], []);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_relayProtocol.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_relayProtocol.html
new file mode 100644
index 0000000000..cdc328fd2b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_relayProtocol.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1435789",
+  title: "WebRTC local-candidate relayProtocol stats attribute"
+});
+
+// This test uses the NAT simulator in order to get srflx candidates.
+// It doesn't work in https, so we turn on getUserMedia in http, which requires
+// a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.nat_simulator.filtering_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.getusermedia.insecure.enabled', true]);
+    const test = new PeerConnectionTest(options);
+    makeOffererNonTrickle(test.chain);
+    makeAnswererNonTrickle(test.chain);
+
+    test.chain.removeAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW");
+    test.chain.append([PC_LOCAL_TEST_LOCAL_STATS_RELAYCANDIDATE]);
+
+    test.setMediaConstraints([{ audio: true }], [{ audio: true }]);
+    await test.run();
+  }, { useIceServer: true });
+}
+
+const PC_LOCAL_TEST_LOCAL_STATS_RELAYCANDIDATE = test => {
+  return test.pcLocal.getStats().then(stats => {
+    let haveRelayProtocol = {};
+    for (let [k, v] of stats) {
+      if (v.type == "local-candidate") {
+        haveRelayProtocol[v.candidateType + "-" + v.relayProtocol] = v.relayProtocol;
+      }
+    }
+    is(haveRelayProtocol["host-undefined"], undefined, "relayProtocol not set for host candidates");
+    is(haveRelayProtocol["srflx-undefined"], undefined, "relayProtocol not set for server reflexive candidates");
+    ok(haveRelayProtocol["relay-udp"], "Has UDP relay candidate");
+    ok(haveRelayProtocol["relay-tcp"], "Has TCP relay candidate");
+    ok(haveRelayProtocol["relay-tls"], "Has TLS relay candidate");
+    is(Object.keys(haveRelayProtocol).length, 5, "All candidate types are accounted for");
+  });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_stereoFmtpPref.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_stereoFmtpPref.html
new file mode 100644
index 0000000000..ab7811fe82
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_stereoFmtpPref.html
@@ -0,0 +1,61 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1793776",
+    title: "Test that media.peerconnection.sdp.disable_stereo_fmtp works"
+  });
+
+  const tests = [
+    async function testStereo() {
+      const offerer = new RTCPeerConnection();
+      offerer.addTransceiver('audio');
+      const answerer = new RTCPeerConnection();
+      await offerer.setLocalDescription();
+      ok(offerer.localDescription.sdp.includes('stereo=1'),
+          'Offer uses stereo=1 when media.peerconnection.sdp.disable_stereo_fmtp is not set');
+      await answerer.setRemoteDescription(offerer.localDescription);
+      const {sdp} = await answerer.createAnswer();
+      ok(sdp.includes('stereo=1'), 'Answer uses stereo=1 when media.peerconnection.sdp.disable_stereo_fmtp is not set');
+    },
+
+    async function testNoStereo() {
+      await pushPrefs(
+          ['media.peerconnection.sdp.disable_stereo_fmtp', true]);
+
+      const offerer = new RTCPeerConnection();
+      offerer.addTransceiver('audio');
+      const answerer = new RTCPeerConnection();
+      await offerer.setLocalDescription();
+      ok(offerer.localDescription.sdp.includes('stereo=0'),
+          'Offer uses stereo=0 when media.peerconnection.sdp.disable_stereo_fmtp is set');
+      await answerer.setRemoteDescription(offerer.localDescription);
+      const {sdp} = await answerer.createAnswer();
+      ok(sdp.includes('stereo=0'), 'Answer uses stereo=0 when media.peerconnection.sdp.disable_stereo_fmtp is set');
+    },
+  ];
+
+  runNetworkTest(async () => {
+    for (const test of tests) {
+      info(`Running test: ${test.name}`);
+      try {
+        await test();
+      } catch (e) {
+        ok(false, `Caught ${e.name}: ${e.message} ${e.stack}`);
+      }
+      info(`Done running test: ${test.name}`);
+      // Make sure we don't build up a pile of GC work, and also get PCImpl to
+      // print their timecards.
+      await new Promise(r => SpecialPowers.exactGC(r));
+    }
+
+    await SpecialPowers.popPrefEnv();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_syncSetDescription.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_syncSetDescription.html
new file mode 100644
index 0000000000..98f0de1b4a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_syncSetDescription.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1063971",
+    title: "Legacy sync setDescription calls",
+    visible: true
+  });
+
+// Test setDescription without callbacks, which many webrtc examples still do
+
+function PC_LOCAL_SET_LOCAL_DESCRIPTION_SYNC(test) {
+  test.pcLocal.onsignalingstatechange = function() {};
+  test.pcLocal._pc.setLocalDescription(test.originalOffer);
+}
+
+function PC_REMOTE_SET_REMOTE_DESCRIPTION_SYNC(test) {
+  test.pcRemote.onsignalingstatechange = function() {};
+  test.pcRemote._pc.setRemoteDescription(test._local_offer,
+    test.pcRemote.releaseIceCandidates,
+    generateErrorCallback("pcRemote._pc.setRemoteDescription() sync failed"));
+}
+function PC_REMOTE_SET_LOCAL_DESCRIPTION_SYNC(test) {
+  test.pcRemote.onsignalingstatechange = function() {};
+  test.pcRemote._pc.setLocalDescription(test.originalAnswer);
+}
+function PC_LOCAL_SET_REMOTE_DESCRIPTION_SYNC(test) {
+  test.pcLocal.onsignalingstatechange = function() {};
+  test.pcLocal._pc.setRemoteDescription(test._remote_answer,
+    test.pcLocal.releaseIceCandidates,
+    generateErrorCallback("pcLocal._pc.setRemoteDescription() sync failed"));
+}
+
+runNetworkTest(() => {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{video: true}], [{video: true}]);
+  test.chain.replace("PC_LOCAL_SET_LOCAL_DESCRIPTION", PC_LOCAL_SET_LOCAL_DESCRIPTION_SYNC);
+  test.chain.replace("PC_REMOTE_SET_REMOTE_DESCRIPTION", PC_REMOTE_SET_REMOTE_DESCRIPTION_SYNC);
+  test.chain.remove("PC_REMOTE_CHECK_CAN_TRICKLE_SYNC");
+  test.chain.replace("PC_REMOTE_SET_LOCAL_DESCRIPTION", PC_REMOTE_SET_LOCAL_DESCRIPTION_SYNC);
+  test.chain.replace("PC_LOCAL_SET_REMOTE_DESCRIPTION", PC_LOCAL_SET_REMOTE_DESCRIPTION_SYNC);
+  test.chain.remove("PC_LOCAL_CHECK_CAN_TRICKLE_SYNC");
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_telephoneEventFirst.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_telephoneEventFirst.html
new file mode 100644
index 0000000000..bde51c1fd0
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_telephoneEventFirst.html
@@ -0,0 +1,56 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "RTCPeerConnection with telephone-event codec first in SDP",
+  bug: "1581898",
+  visible: true
+});
+
+const test = async () => {
+  const pc1 = new RTCPeerConnection();
+  const pc2 = new RTCPeerConnection();
+  const stream = await navigator.mediaDevices.getUserMedia({audio:true});
+  pc1.addTrack(stream.getAudioTracks()[0], stream);
+  pc2.addTrack(stream.getAudioTracks()[0], stream);
+
+  const offer = await pc1.createOffer();
+  await pc1.setLocalDescription(offer);
+
+  const regex = /^(m=audio \d+ [^ ]+) (.*) 101(.*)$/m;
+
+  // Rewrite offer so payload type 101 comes first
+  offer.sdp = offer.sdp.replace(regex, '$1 101 $2 $3');
+
+  ok(offer.sdp.match(/^m=audio \d+ [^ ]+ 101 /m),
+     "Payload type 101 should be first on the m-line");
+
+  await pc2.setRemoteDescription(offer);
+  const answer = await pc2.createAnswer();
+
+  pc1.onicecandidate = e => { pc2.addIceCandidate(e.candidate); }
+  pc2.onicecandidate = e => { pc1.addIceCandidate(e.candidate); }
+
+  await pc1.setRemoteDescription(answer);
+  await pc2.setLocalDescription(answer);
+  await new Promise(resolve => {
+    pc1.oniceconnectionstatechange = e => {
+      if (pc1.iceConnectionState == "connected") {
+        resolve();
+      }
+    };
+  });
+  await wait(1000);
+};
+
+runNetworkTest(test);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_threeUnbundledConnections.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_threeUnbundledConnections.html
new file mode 100644
index 0000000000..75f0d12463
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_threeUnbundledConnections.html
@@ -0,0 +1,134 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1342579",
+    title: "Unbundled PC connects to two different PCs",
+    visible: true
+  });
+
+  const fakeFingerPrint = "a=fingerprint:sha-256 11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11";
+
+  const pc1 = new RTCPeerConnection();
+  const pc2 = new RTCPeerConnection();
+  const pc3 = new RTCPeerConnection();
+
+  const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+  pc1.onicecandidate = e => {
+    if (e.candidate) {
+      if (e.candidate.sdpMid === "1") {
+        add(pc2, e.candidate, generateErrorCallback())
+      } else {
+        add(pc3, e.candidate, generateErrorCallback())
+      }
+    }
+  };
+  pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+  pc3.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+  let ice1Finished, ice2Finished, ice3Finished;
+  const ice1Done = new Promise(r => ice1Finished = r);
+  const ice2Done = new Promise(r => ice2Finished = r);
+  const ice3Done = new Promise(r => ice3Finished = r);
+
+  const icsc = (pc, str, resolve) => {
+    const state = pc.iceConnectionState;
+    info(str + " ICE connection state is: " + state);
+    if (state == "connected") {
+      ok(true, str + " ICE connected");
+      resolve();
+    } else if (state == "failed") {
+      ok(false, str + " ICE failed")
+      resolve();
+    }
+  };
+
+  pc1.oniceconnectionstatechange = e => icsc(pc1, "PC1", ice1Finished);
+  pc2.oniceconnectionstatechange = e => icsc(pc2, "PC2", ice2Finished);
+  pc3.oniceconnectionstatechange = e => icsc(pc3, "PC3", ice3Finished);
+
+
+  function combineAnswer(origAnswer, answer) {
+    const sdplines = origAnswer.sdp.split('\r\n');
+    const fpIndex = sdplines.findIndex(l => l.match('^a=fingerprint'));
+    const FP = sdplines[fpIndex];
+    const audioIndex = sdplines.findIndex(l => l.match(/^m=audio [1-9]/));
+    const videoIndex = sdplines.findIndex(l => l.match(/^m=video [1-9]/));
+    if (audioIndex > -1) {
+      var ss = sdplines.slice(0, audioIndex);
+      ss.splice(fpIndex, 1);
+      answer.sessionSection = ss;
+      const rejectedVideoIndex = sdplines.findIndex(l => l.match('m=video 0'));
+      var ams = sdplines.slice(audioIndex, rejectedVideoIndex);
+      ams.push(FP);
+      ams.push(fakeFingerPrint);
+      answer.audioMsection = ams;
+    }
+    if (videoIndex > -1) {
+      var vms = sdplines.slice(videoIndex, sdplines.length -1);
+      vms.push(fakeFingerPrint);
+      vms.push(FP);
+      answer.videoMsection = vms;
+    }
+    return answer;
+  }
+
+runNetworkTest(async () => {
+    const v1 = createMediaElement('video', 'v1');
+    const v2 = createMediaElement('video', 'v2');
+    const v3 = createMediaElement('video', 'v3');
+
+    const stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
+    (v1.srcObject = stream).getTracks().forEach(t => pc1.addTrack(t, stream));
+
+    const stream2 = await navigator.mediaDevices.getUserMedia({ video: true });
+    (v2.srcObject = stream2).getTracks().forEach(t => pc2.addTrack(t, stream2));
+
+    const stream3 = await navigator.mediaDevices.getUserMedia({ audio: true });
+    (v3.srcObject = stream3).getTracks().forEach(t => pc3.addTrack(t, stream3));
+
+    const offer = await pc1.createOffer();
+    await pc1.setLocalDescription(offer);
+
+    //info("Original OFFER: " + JSON.stringify(offer));
+    offer.sdp = sdputils.removeBundle(offer.sdp);
+    //info("OFFER w/o BUNDLE: " + JSON.stringify(offer));
+    const offerAudio = new RTCSessionDescription(JSON.parse(JSON.stringify(offer)));
+    offerAudio.sdp = offerAudio.sdp.replace('m=video 9', 'm=video 0');
+    //info("offerAudio: " + JSON.stringify(offerAudio));
+    const offerVideo = new RTCSessionDescription(JSON.parse(JSON.stringify(offer)));
+    offerVideo.sdp = offerVideo.sdp.replace('m=audio 9', 'm=audio 0');
+    //info("offerVideo: " + JSON.stringify(offerVideo));
+
+    // We need to do these in parallel, otherwise pc1 will start firing
+    // icecandidate events before pc3 is ready.
+    await Promise.all([pc2.setRemoteDescription(offerVideo), pc3.setRemoteDescription(offerAudio)]);
+
+    const answerVideo = await pc2.createAnswer();
+    const answerAudio = await pc3.createAnswer();
+
+    const answer = combineAnswer(answerAudio, combineAnswer(answerVideo, {}));
+    const fakeAnswer = answer.sessionSection.concat(answer.audioMsection, answer.videoMsection).join('\r\n');
+    info("ANSWER: " + fakeAnswer);
+
+    // We want to do these in parallel, because if we do them seqentially, by the
+    // time pc3.sLD completes pc2 could have fired icecandidate events, when we
+    // haven't called pc1.sRD yet.
+    await Promise.all(
+      [pc2.setLocalDescription(answerVideo),
+       pc3.setLocalDescription(answerAudio),
+       pc1.setRemoteDescription({type: 'answer', sdp: fakeAnswer})]);
+
+    await Promise.all([ice1Done, ice2Done, ice3Done]);
+
+    ok(true, "Connected.");
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_throwInCallbacks.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_throwInCallbacks.html
new file mode 100644
index 0000000000..5a3872c120
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_throwInCallbacks.html
@@ -0,0 +1,83 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "857765",
+    title: "Throw in PeerConnection callbacks"
+  });
+
+runNetworkTest(function () {
+  let finish;
+  const onfinished = new Promise(r => finish = async () => {
+    window.onerror = oldOnError;
+    is(error_count, 7, "Seven expected errors verified.");
+    r();
+  });
+
+  function getFail() {
+    return err => {
+      window.onerror = oldOnError;
+      generateErrorCallback()(err);
+    };
+  }
+
+  let error_count = 0;
+  let oldOnError = window.onerror;
+  window.onerror = (errorMsg, url, lineNumber) => {
+    if (!errorMsg.includes("Expected")) {
+      getFail()(errorMsg);
+    }
+    error_count += 1;
+    info("onerror " + error_count + ": " + errorMsg);
+    if (error_count == 7) {
+      finish();
+    }
+    throw new Error("window.onerror may throw");
+    return false;
+  }
+
+  let pc0, pc1, pc2;
+  // Test failure callbacks (limited to 1 for now)
+  pc0 = new RTCPeerConnection();
+  pc0.close();
+  pc0.createOffer(getFail(), function(err) {
+    pc1 = new RTCPeerConnection();
+    pc2 = new RTCPeerConnection();
+
+    // Test success callbacks (happy path)
+    navigator.mozGetUserMedia({video:true}, function(video1) {
+      pc1.addStream(video1);
+      pc1.createOffer(function(offer) {
+        pc1.setLocalDescription(offer, function() {
+          pc2.setRemoteDescription(offer, function() {
+            pc2.createAnswer(function(answer) {
+              pc2.setLocalDescription(answer, function() {
+                pc1.setRemoteDescription(answer, function() {
+                  throw new Error("Expected");
+                }, getFail());
+                throw new Error("Expected");
+              }, getFail());
+              throw new Error("Expected");
+            }, getFail());
+            throw new Error("Expected");
+          }, getFail());
+          throw new Error("Expected");
+        }, getFail());
+        throw new Error("Expected");
+      }, getFail());
+    }, getFail());
+    throw new Error("Expected");
+  });
+
+  return onfinished;
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_toJSON.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_toJSON.html
new file mode 100644
index 0000000000..96c2c42b78
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_toJSON.html
@@ -0,0 +1,39 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "928304",
+    title: "test toJSON() on RTCSessionDescription and RTCIceCandidate"
+  });
+
+  runNetworkTest(function () {
+    /** Test for Bug 872377 **/
+
+    var rtcSession = new RTCSessionDescription({ sdp: "Picklechips!",
+                                                 type: "offer" });
+    var jsonCopy = JSON.parse(JSON.stringify(rtcSession));
+    for (key in rtcSession) {
+      if (typeof(rtcSession[key]) == "function") continue;
+      is(rtcSession[key], jsonCopy[key], "key " + key + " should match.");
+    }
+
+    /** Test for Bug 928304 **/
+
+    var rtcIceCandidate = new RTCIceCandidate({ candidate: "dummy",
+                                                   sdpMid: "test",
+                                                   sdpMLineIndex: 3 });
+    jsonCopy = JSON.parse(JSON.stringify(rtcIceCandidate));
+    for (key in rtcIceCandidate) {
+      if (typeof(rtcIceCandidate[key]) == "function") continue;
+      is(rtcIceCandidate[key], jsonCopy[key], "key " + key + " should match.");
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_trackDisabling.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_trackDisabling.html
new file mode 100644
index 0000000000..73323cf007
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_trackDisabling.html
@@ -0,0 +1,108 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1219711",
+  title: "Disabling locally should be reflected remotely"
+});
+
+runNetworkTest(async () => {
+  const test = new PeerConnectionTest();
+
+  await pushPrefs(
+    ["media.getusermedia.camera.stop_on_disable.enabled", true],
+    ["media.getusermedia.camera.stop_on_disable.delay_ms", 0],
+    ["media.getusermedia.microphone.stop_on_disable.enabled", true],
+    ["media.getusermedia.microphone.stop_on_disable.delay_ms", 0],
+    // Always use fake tracks since we depend on video to be somewhat green and
+    // audio to have a large 1000Hz component (or 440Hz if using fake devices).
+    ['media.audio_loopback_dev', ''],
+    ['media.video_loopback_dev', ''],
+    ['media.navigator.streams.fake', true]);
+
+  test.setMediaConstraints([{audio: true, video: true}], []);
+  test.chain.append([
+    function CHECK_ASSUMPTIONS() {
+      is(test.pcLocal.localMediaElements.length, 2,
+         "pcLocal should have one media element");
+      is(test.pcRemote.remoteMediaElements.length, 2,
+         "pcRemote should have one media element");
+      is(test.pcLocal._pc.getLocalStreams().length, 1,
+         "pcLocal should have one stream");
+      is(test.pcRemote._pc.getRemoteStreams().length, 1,
+         "pcRemote should have one stream");
+    },
+    async function CHECK_VIDEO() {
+      const h = new CaptureStreamTestHelper2D();
+      const localVideo = test.pcLocal.localMediaElements
+        .find(e => e instanceof HTMLVideoElement);
+      const remoteVideo = test.pcRemote.remoteMediaElements
+        .find(e => e instanceof HTMLVideoElement);
+      // We check a pixel somewhere away from the top left corner since
+      // MediaEngineFake puts semi-transparent time indicators there.
+      const offsetX = 50;
+      const offsetY = 50;
+      const threshold = 128;
+
+      // We're regarding black as disabled here, and we're setting the alpha
+      // channel of the pixel to 255 to disregard alpha when testing.
+      const checkVideoEnabled = video => h.waitForPixel(video,
+        px => (px[3] = 255, h.isPixelNot(px, h.black, threshold)),
+        { offsetX, offsetY }
+      );
+      const checkVideoDisabled = video => h.waitForPixel(video,
+        px => (px[3] = 255, h.isPixel(px, h.black, threshold)),
+        { offsetX, offsetY }
+      );
+
+      info("Checking local video enabled");
+      await checkVideoEnabled(localVideo);
+      info("Checking remote video enabled");
+      await checkVideoEnabled(remoteVideo);
+
+      info("Disabling original");
+      test.pcLocal._pc.getLocalStreams()[0].getVideoTracks()[0].enabled = false;
+
+      info("Checking local video disabled");
+      await checkVideoDisabled(localVideo);
+      info("Checking remote video disabled");
+      await checkVideoDisabled(remoteVideo);
+    },
+    async function CHECK_AUDIO() {
+      const ac = new AudioContext();
+      const localAnalyser = new AudioStreamAnalyser(ac, test.pcLocal._pc.getLocalStreams()[0]);
+      const remoteAnalyser = new AudioStreamAnalyser(ac, test.pcRemote._pc.getRemoteStreams()[0]);
+
+      const checkAudio = (analyser, fun) => analyser.waitForAnalysisSuccess(fun);
+
+      const freq = localAnalyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+      const checkAudioEnabled = analyser =>
+        checkAudio(analyser, array => array[freq] > 200);
+      const checkAudioDisabled = analyser =>
+        checkAudio(analyser, array => array[freq] < 50);
+
+      info("Checking local audio enabled");
+      await checkAudioEnabled(localAnalyser);
+      info("Checking remote audio enabled");
+      await checkAudioEnabled(remoteAnalyser);
+
+      test.pcLocal._pc.getLocalStreams()[0].getAudioTracks()[0].enabled = false;
+
+      info("Checking local audio disabled");
+      await checkAudioDisabled(localAnalyser);
+      info("Checking remote audio disabled");
+      await checkAudioDisabled(remoteAnalyser);
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_trackDisabling_clones.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_trackDisabling_clones.html
new file mode 100644
index 0000000000..ae7647fa1a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_trackDisabling_clones.html
@@ -0,0 +1,162 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1219711",
+  title: "Disabling locally should be reflected remotely, individually for clones"
+});
+
+runNetworkTest(async () => {
+  var test = new PeerConnectionTest();
+
+  await pushPrefs(
+    ["media.getusermedia.camera.stop_on_disable.enabled", true],
+    ["media.getusermedia.camera.stop_on_disable.delay_ms", 0],
+    ["media.getusermedia.microphone.stop_on_disable.enabled", true],
+    ["media.getusermedia.microphone.stop_on_disable.delay_ms", 0],
+    // Always use fake tracks since we depend on audio to have a large 1000Hz
+    // component.
+    ['media.audio_loopback_dev', ''],
+    ['media.navigator.streams.fake', true]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+  var originalStream;
+  var localVideoOriginal;
+
+  test.setMediaConstraints([{audio: true, video: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function PC_LOCAL_GUM_CLONE() {
+      return getUserMedia(test.pcLocal.constraints[0]).then(stream => {
+        originalStream = stream;
+        localVideoOriginal =
+          createMediaElement("video", "local-original");
+        localVideoOriginal.srcObject = stream;
+        test.pcLocal.attachLocalStream(originalStream.clone());
+      });
+    }
+  ]);
+  test.chain.append([
+    function CHECK_ASSUMPTIONS() {
+      is(test.pcLocal.localMediaElements.length, 2,
+         "pcLocal should have one media element");
+      is(test.pcRemote.remoteMediaElements.length, 2,
+         "pcRemote should have one media element");
+      is(test.pcLocal._pc.getLocalStreams().length, 1,
+         "pcLocal should have one stream");
+      is(test.pcRemote._pc.getRemoteStreams().length, 1,
+         "pcRemote should have one stream");
+    },
+    async function CHECK_VIDEO() {
+      info("Checking video");
+      var h = new CaptureStreamTestHelper2D();
+      var localVideoClone = test.pcLocal.localMediaElements
+        .find(e => e instanceof HTMLVideoElement);
+      var remoteVideoClone = test.pcRemote.remoteMediaElements
+        .find(e => e instanceof HTMLVideoElement);
+
+      // We check a pixel somewhere away from the top left corner since
+      // MediaEngineFake puts semi-transparent time indicators there.
+      const offsetX = 50;
+      const offsetY = 50;
+      const threshold = 128;
+      const remoteDisabledColor = h.black;
+
+      // We're regarding black as disabled here, and we're setting the alpha
+      // channel of the pixel to 255 to disregard alpha when testing.
+      var checkVideoEnabled = video => h.waitForPixel(video,
+        px => (px[3] = 255, h.isPixelNot(px, h.black, threshold)),
+        { offsetX, offsetY }
+      );
+      var checkVideoDisabled = video => h.waitForPixel(video,
+        px => (px[3] = 255, h.isPixel(px, h.black, threshold)),
+        { offsetX, offsetY }
+      );
+
+      info("Checking local original enabled");
+      await checkVideoEnabled(localVideoOriginal);
+      info("Checking local clone enabled");
+      await checkVideoEnabled(localVideoClone);
+      info("Checking remote clone enabled");
+      await checkVideoEnabled(remoteVideoClone);
+
+      info("Disabling original");
+      originalStream.getVideoTracks()[0].enabled = false;
+
+      info("Checking local original disabled");
+      await checkVideoDisabled(localVideoOriginal);
+      info("Checking local clone enabled");
+      await checkVideoEnabled(localVideoClone);
+      info("Checking remote clone enabled");
+      await checkVideoEnabled(remoteVideoClone);
+
+      info("Re-enabling original; disabling clone");
+      originalStream.getVideoTracks()[0].enabled = true;
+      test.pcLocal._pc.getLocalStreams()[0].getVideoTracks()[0].enabled = false;
+
+      info("Checking local original enabled");
+      await checkVideoEnabled(localVideoOriginal);
+      info("Checking local clone disabled");
+      await checkVideoDisabled(localVideoClone);
+      info("Checking remote clone disabled");
+      await checkVideoDisabled(remoteVideoClone);
+    },
+    async function CHECK_AUDIO() {
+      info("Checking audio");
+      var ac = new AudioContext();
+      var localAnalyserOriginal = new AudioStreamAnalyser(ac, originalStream);
+      var localAnalyserClone =
+        new AudioStreamAnalyser(ac, test.pcLocal._pc.getLocalStreams()[0]);
+      var remoteAnalyserClone =
+        new AudioStreamAnalyser(ac, test.pcRemote._pc.getRemoteStreams()[0]);
+
+      var freq = localAnalyserOriginal.binIndexForFrequency(TEST_AUDIO_FREQ);
+      var checkAudioEnabled = analyser =>
+        analyser.waitForAnalysisSuccess(array => array[freq] > 200);
+      var checkAudioDisabled = analyser =>
+        analyser.waitForAnalysisSuccess(array => array[freq] < 50);
+
+      info("Checking local original enabled");
+      await checkAudioEnabled(localAnalyserOriginal);
+      info("Checking local clone enabled");
+      await checkAudioEnabled(localAnalyserClone);
+      info("Checking remote clone enabled");
+      await checkAudioEnabled(remoteAnalyserClone);
+
+      info("Disabling original");
+      originalStream.getAudioTracks()[0].enabled = false;
+
+      info("Checking local original disabled");
+      await checkAudioDisabled(localAnalyserOriginal);
+      info("Checking local clone enabled");
+      await checkAudioEnabled(localAnalyserClone);
+      info("Checking remote clone enabled");
+      await checkAudioEnabled(remoteAnalyserClone);
+
+      info("Re-enabling original; disabling clone");
+      originalStream.getAudioTracks()[0].enabled = true;
+      test.pcLocal._pc.getLocalStreams()[0].getAudioTracks()[0].enabled = false;
+
+      info("Checking local original enabled");
+      await checkAudioEnabled(localAnalyserOriginal);
+      info("Checking local clone disabled");
+      await checkAudioDisabled(localAnalyserClone);
+      info("Checking remote clone disabled");
+      await checkAudioDisabled(remoteAnalyserClone);
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_trackless_sender_stats.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_trackless_sender_stats.html
new file mode 100644
index 0000000000..f0356f5655
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_trackless_sender_stats.html
@@ -0,0 +1,56 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1452673",
+    title: "Trackless RTCRtpSender.getStats()",
+    visible: true
+  });
+
+  // Calling getstats() on a trackless RTCRtpSender should yield an empty
+  // stats report.  When track stats are added in the future, the stats
+  // for the removed tracks should continue to appear.
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.chain.removeAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW");
+    test.chain.append(
+      async function PC_LOCAL_AND_REMOTE_TRACKLESS_SENDER_STATS(test) {
+        await Promise.all([
+          waitForSyncedRtcp(test.pcLocal._pc),
+          waitForSyncedRtcp(test.pcRemote._pc),
+        ]);
+        let senders = test.pcLocal.getSenders();
+        let receivers = test.pcRemote.getReceivers();
+        is(senders.length, 2, "Have exactly two senders.");
+        is(receivers.length, 2, "Have exactly two receivers.");
+        for(let kind of ["audio", "video"]) {
+          is(senders.filter(s => s.track.kind == kind).length, 1,
+              "Exactly 1 sender of kind " + kind);
+          is(receivers.filter(r => r.track.kind == kind).length, 1,
+              "Exactly 1 receiver of kind " + kind);
+        }
+        // Remove tracks from senders
+        for (const sender of senders) {
+          await sender.replaceTrack(null);
+          is(sender.track, null, "Sender track removed");
+          let stats = await sender.getStats();
+          ok(stats instanceof window.RTCStatsReport, "Stats is instance of RTCStatsReport");
+          // Number of stats in the report. This should be 0.
+          is(stats.size, 0, "Trackless sender stats report is empty");
+        }
+      }
+    );
+    test.setMediaConstraints([{audio: true}, {video: true}], []);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioStreams.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioStreams.html
new file mode 100644
index 0000000000..7ea18ab3dd
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioStreams.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1091242",
+    title: "Multistream: Two audio streams"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {audio: true}],
+                             [{audio: true}, {audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioTracksInOneStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioTracksInOneStream.html
new file mode 100644
index 0000000000..99d4ad625a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioTracksInOneStream.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1145407",
+    title: "Multistream: Two audio tracks in one stream"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.chain.insertAfter("PC_REMOTE_GET_OFFER", [
+        function PC_REMOTE_OVERRIDE_STREAM_IDS_IN_OFFER(test) {
+          test._local_offer.sdp = test._local_offer.sdp.replace(
+              /a=msid:[^\s]*/g,
+              "a=msid:foo");
+        }
+    ]);
+    test.chain.insertAfter("PC_LOCAL_GET_ANSWER", [
+        function PC_LOCAL_OVERRIDE_STREAM_IDS_IN_ANSWER(test) {
+          test._remote_answer.sdp = test._remote_answer.sdp.replace(
+              /a=msid:[^\s]*/g,
+              "a=msid:foo");
+        }
+    ]);
+    test.setMediaConstraints([{audio: true}, {audio: true}],
+                             [{audio: true}, {audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreams.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreams.html
new file mode 100644
index 0000000000..5f4bd463d4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreams.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+
+  createHTML({
+    bug: "1091242",
+    title: "Multistream: Two audio streams, two video streams"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}, {audio: true},
+                                {video: true}],
+                             [{audio: true}, {video: true}, {audio: true},
+                                {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreamsCombined.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreamsCombined.html
new file mode 100644
index 0000000000..fcc9c6c8fa
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreamsCombined.html
@@ -0,0 +1,70 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+
+  createHTML({
+    bug: "1091242",
+    title: "Multistream: Two audio/video streams"
+  });
+
+  runNetworkTest(async (options) => {
+    // Disable platform encodre for SW MFT encoder causes some stats
+    // exceeding the test thresholds.
+    // E.g. inbound-rtp.packetsDiscarded value=118 >= 100.
+    await matchPlatformH264CodecPrefs();
+
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true, video: true},
+                                {audio: true, video: true}],
+                             [{audio: true, video: true},
+                                {audio: true, video: true}]);
+
+    // Test stats, including coalescing of codec stats.
+    test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+      [PC_LOCAL_TEST_LOCAL_STATS]);
+
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      [PC_REMOTE_TEST_REMOTE_STATS]);
+
+    const testCoalescedCodecStats = stats => {
+      is([...stats.values()].filter(({type}) => type.endsWith("rtp")).length,
+        16,
+        "Expected: 4 outbound, 4 remote-inbound, 4 inbound, 4 remote-inbound");
+      const codecs = [...stats.values()]
+          .filter(({type}) => type == "codec")
+          .sort((a, b) => a.mimeType > b.mimeType);
+      is(codecs.length, 2, "Should have registered two codecs (coalesced)");
+      is(new Set(codecs.map(({transportId}) => transportId)).size, 1,
+        "Should have registered only one transport with BUNDLE");
+      const codecTypes = new Set(codecs.map(({codecType}) => codecType));
+      is(codecTypes.size, 1,
+        "Should have identical encode and decode configurations (and stats)");
+      is(codecTypes[0], undefined,
+        "Should have identical encode and decode configurations (and stats)");
+      is(codecs[0].mimeType.slice(0, 5), "audio",
+        "Should have registered an audio codec");
+      is(codecs[1].mimeType.slice(0, 5), "video",
+        "Should have registered a video codec");
+    };
+
+    test.chain.append([
+      async function PC_LOCAL_TEST_COALESCED_CODEC_STATS() {
+        testCoalescedCodecStats(await test.pcLocal._pc.getStats());
+      },
+      async function PC_REMOTE_TEST_COALESCED_CODEC_STATS() {
+        testCoalescedCodecStats(await test.pcRemote._pc.getStats());
+      },
+    ]);
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreamsCombinedNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreamsCombinedNoBundle.html
new file mode 100644
index 0000000000..8b825db617
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreamsCombinedNoBundle.html
@@ -0,0 +1,107 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1225722",
+  title: "Multistream: Two audio/video streams without BUNDLE"
+});
+
+runNetworkTest(async (options = {}) => {
+  // Disable platform encodre for SW MFT encoder causes some stats
+  // exceeding the test thresholds.
+  // E.g. inbound-rtp.packetsDiscarded value=118 >= 100.
+  await matchPlatformH264CodecPrefs();
+
+  options.bundle = false;
+  const test = new PeerConnectionTest(options);
+  test.setMediaConstraints(
+    [{audio: true, video: true}, {audio: true, video: true}],
+    [{audio: true, video: true}, {audio: true, video: true}]
+  );
+
+  // Test stats, including that codec stats do not coalesce without BUNDLE.
+  const testNonBundledStats = async pc => {
+    // This is basically PC_*_TEST_*_STATS fleshed out, but uses
+    // sender/receiver.getStats instead of pc.getStats, since the codec stats
+    // code assumes at most one sender and at most one receiver.
+    await waitForSyncedRtcp(pc);
+    const senderPromises = pc.getSenders().map(obj => obj.getStats());
+    const receiverPromises = pc.getReceivers().map(obj => obj.getStats());
+    const senderStats = await Promise.all(senderPromises);
+    const receiverStats = await Promise.all(receiverPromises);
+    for (const stats of [...senderStats, ...receiverStats]) {
+      checkExpectedFields(stats);
+      pedanticChecks(stats);
+    }
+    for (const stats of senderStats) {
+      checkSenderStats(stats, 1);
+    }
+  };
+
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW", [
+    async function PC_LOCAL_TEST_LOCAL_NONBUNDLED_STATS(test) {
+      await testNonBundledStats(test.pcLocal._pc);
+    },
+  ]);
+
+  test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW", [
+    async function PC_REMOTE_TEST_LOCAL_NONBUNDLED_STATS(test) {
+      await testNonBundledStats(test.pcRemote._pc);
+    },
+  ]);
+
+  const testNonCoalescedCodecStats = stats => {
+    const codecs = [...stats.values()]
+          .filter(({type}) => type == "codec");
+    is([...stats.values()].filter(({type}) => type.endsWith("rtp")).length, 16,
+      "Expected: 4 outbound, 4 remote-inbound, 4 inbound, 4 remote-inbound");
+    const codecTypes = new Set(codecs.map(({codecType}) => codecType));
+    is(codecTypes.size, 1,
+      "Should have identical encode and decode configurations (and stats)");
+    is(codecTypes[0], undefined,
+      "Should have identical encode and decode configurations (and stats)");
+    const transportIds = new Set(codecs.map(({transportId}) => transportId));
+    is(transportIds.size, 4,
+      "Should have registered four transports for two sendrecv streams");
+    for (const transportId of transportIds) {
+      is(codecs.filter(c => c.transportId == transportId).length, 1,
+        "Should have registered one codec per transport without BUNDLE");
+    }
+    for (const prefix of ["audio", "video"]) {
+      const prefixed = codecs.filter(c => c.mimeType.startsWith(prefix));
+      is(prefixed.length, 2, `Should have registered two ${prefix} codecs`);
+      if (prefixed.length == 2) {
+        is(prefixed[0].payloadType, prefixed[1].payloadType,
+          "same payloadType");
+        isnot(prefixed[0].transportId, prefixed[1].transportId,
+          "different transportIds");
+        is(prefixed[0].mimeType, prefixed[1].mimeType, "same mimeType");
+        is(prefixed[0].clockRate, prefixed[1].clockRate, "same clockRate");
+        is(prefixed[0].channels, prefixed[1].channels, "same channels");
+        is(prefixed[0].sdpFmtpLine, prefixed[1].sdpFmtpLine,
+          "same sdpFmtpLine");
+      }
+    }
+  };
+
+  test.chain.append([
+    async function PC_LOCAL_TEST_NON_COALESCED_CODEC_STATS() {
+      testNonCoalescedCodecStats(await test.pcLocal._pc.getStats());
+    },
+    async function PC_REMOTE_TEST_NON_COALESCED_CODEC_STATS() {
+      testNonCoalescedCodecStats(await test.pcRemote._pc.getStats());
+    },
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoVideoStreams.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoVideoStreams.html
new file mode 100644
index 0000000000..0ab180cc55
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoVideoStreams.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1091242",
+    title: "Multistream: Two video streams"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}, {video: true}],
+                             [{video: true}, {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoVideoTracksInOneStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoVideoTracksInOneStream.html
new file mode 100644
index 0000000000..4eaf8b3f48
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoVideoTracksInOneStream.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1145407",
+    title: "Multistream: Two video tracks in offerer stream"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.chain.insertAfter("PC_REMOTE_GET_OFFER", [
+        function PC_REMOTE_OVERRIDE_STREAM_IDS_IN_OFFER(test) {
+          test._local_offer.sdp = test._local_offer.sdp.replace(
+              /a=msid:[^\s]*/g,
+              "a=msid:foo");
+        }
+    ]);
+    test.chain.insertAfter("PC_LOCAL_GET_ANSWER", [
+        function PC_LOCAL_OVERRIDE_STREAM_IDS_IN_ANSWER(test) {
+          test._remote_answer.sdp = test._remote_answer.sdp.replace(
+              /a=msid:[^\s]*/g,
+              "a=msid:foo");
+        }
+    ]);
+    test.setMediaConstraints([{video: true}, {video: true}],
+                             [{video: true}, {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyAudioAfterRenegotiation.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyAudioAfterRenegotiation.html
new file mode 100644
index 0000000000..86ef6d4678
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyAudioAfterRenegotiation.html
@@ -0,0 +1,99 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1166832",
+    title: "Renegotiation: verify audio after renegotiation"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    const helper = new AudioStreamHelper();
+
+    test.chain.append([
+      function CHECK_ASSUMPTIONS() {
+        is(test.pcLocal.localMediaElements.length, 1,
+           "pcLocal should have one media element");
+        is(test.pcRemote.remoteMediaElements.length, 1,
+           "pcRemote should have one media element");
+        is(test.pcLocal._pc.getLocalStreams().length, 1,
+           "pcLocal should have one stream");
+        is(test.pcRemote._pc.getRemoteStreams().length, 1,
+           "pcRemote should have one stream");
+      },
+      function CHECK_AUDIO() {
+        return Promise.resolve()
+          .then(() => info("Checking local audio enabled"))
+          .then(() => helper.checkAudioFlowing(test.pcLocal._pc.getLocalStreams()[0]))
+          .then(() => info("Checking remote audio enabled"))
+          .then(() => helper.checkAudioFlowing(test.pcRemote._pc.getRemoteStreams()[0]))
+
+          .then(() => test.pcLocal._pc.getLocalStreams()[0].getAudioTracks()[0].enabled = false)
+
+          .then(() => info("Checking local audio disabled"))
+          .then(() => helper.checkAudioNotFlowing(test.pcLocal._pc.getLocalStreams()[0]))
+          .then(() => info("Checking remote audio disabled"))
+          .then(() => helper.checkAudioNotFlowing(test.pcRemote._pc.getRemoteStreams()[0]))
+      }
+    ]);
+
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}],
+                                   []);
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+      ]
+    );
+
+    test.chain.append([
+      function CHECK_ASSUMPTIONS2() {
+        is(test.pcLocal.localMediaElements.length, 2,
+           "pcLocal should have two media elements");
+        is(test.pcRemote.remoteMediaElements.length, 2,
+           "pcRemote should have two media elements");
+        is(test.pcLocal._pc.getLocalStreams().length, 2,
+           "pcLocal should have two streams");
+        is(test.pcRemote._pc.getRemoteStreams().length, 2,
+           "pcRemote should have two streams");
+      },
+      function RE_CHECK_AUDIO() {
+        return Promise.resolve()
+          .then(() => info("Checking local audio enabled"))
+          .then(() => helper.checkAudioNotFlowing(test.pcLocal._pc.getLocalStreams()[0]))
+          .then(() => info("Checking remote audio enabled"))
+          .then(() => helper.checkAudioNotFlowing(test.pcRemote._pc.getRemoteStreams()[0]))
+
+          .then(() => info("Checking local2 audio enabled"))
+          .then(() => helper.checkAudioFlowing(test.pcLocal._pc.getLocalStreams()[1]))
+          .then(() => info("Checking remote2 audio enabled"))
+          .then(() => helper.checkAudioFlowing(test.pcRemote._pc.getRemoteStreams()[1]))
+
+          .then(() => test.pcLocal._pc.getLocalStreams()[1].getAudioTracks()[0].enabled = false)
+          .then(() => test.pcLocal._pc.getLocalStreams()[0].getAudioTracks()[0].enabled = true)
+
+          .then(() => info("Checking local2 audio disabled"))
+          .then(() => helper.checkAudioNotFlowing(test.pcLocal._pc.getLocalStreams()[1]))
+          .then(() => info("Checking remote2 audio disabled"))
+          .then(() => helper.checkAudioNotFlowing(test.pcRemote._pc.getRemoteStreams()[1]))
+
+          .then(() => info("Checking local audio enabled"))
+          .then(() => helper.checkAudioFlowing(test.pcLocal._pc.getLocalStreams()[0]))
+          .then(() => info("Checking remote audio enabled"))
+          .then(() => helper.checkAudioFlowing(test.pcRemote._pc.getRemoteStreams()[0]))
+      }
+    ]);
+
+    test.setMediaConstraints([{audio: true}], []);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyDescriptions.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyDescriptions.html
new file mode 100644
index 0000000000..f685f7c99a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyDescriptions.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1264479",
+    title: "PeerConnection verify current and pending descriptions"
+  });
+
+  const pc1 = new RTCPeerConnection();
+  const pc2 = new RTCPeerConnection();
+
+  var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+  pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
+  pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+
+  runNetworkTest(function() {
+    const v1 = createMediaElement('video', 'v1');
+    const v2 = createMediaElement('video', 'v2');
+
+    return navigator.mediaDevices.getUserMedia({ video: true, audio: true })
+    .then(stream => (v1.srcObject = stream).getTracks().forEach(t => pc1.addTrack(t, stream)))
+    .then(() => pc1.createOffer({})) // check that createOffer accepts arg.
+    .then(offer => pc1.setLocalDescription(offer))
+    .then(() => {
+      ok(!pc1.currentLocalDescription, "pc1 currentLocalDescription is empty");
+      ok(pc1.pendingLocalDescription, "pc1 pendingLocalDescription is set");
+      ok(pc1.localDescription, "pc1 localDescription is set");
+    })
+    .then(() => pc2.setRemoteDescription(pc1.localDescription))
+    .then(() => {
+      ok(!pc2.currentRemoteDescription, "pc2 currentRemoteDescription is empty");
+      ok(pc2.pendingRemoteDescription, "pc2 pendingRemoteDescription is set");
+      ok(pc2.remoteDescription, "pc2 remoteDescription is set");
+    })
+    .then(() => pc2.createAnswer({}))  // check that createAnswer accepts arg.
+    .then(answer => pc2.setLocalDescription(answer))
+    .then(() => {
+      ok(pc2.currentLocalDescription, "pc2 currentLocalDescription is set");
+      ok(!pc2.pendingLocalDescription, "pc2 pendingLocalDescription is empty");
+      ok(pc2.localDescription, "pc2 localDescription is set");
+    })
+    .then(() => pc1.setRemoteDescription(pc2.localDescription))
+    .then(() => {
+      ok(pc1.currentRemoteDescription, "pc1 currentRemoteDescription is set");
+      ok(!pc1.pendingRemoteDescription, "pc1 pendingRemoteDescription is empty");
+      ok(pc1.remoteDescription, "pc1 remoteDescription is set");
+    });
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyVideoAfterRenegotiation.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyVideoAfterRenegotiation.html
new file mode 100644
index 0000000000..8d4155ddff
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyVideoAfterRenegotiation.html
@@ -0,0 +1,123 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1166832",
+    title: "Renegotiation: verify video after renegotiation"
+  });
+
+runNetworkTest(async () => {
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  const test = new PeerConnectionTest();
+
+  const h1 = new CaptureStreamTestHelper2D(50, 50);
+  const canvas1 = h1.createAndAppendElement('canvas', 'source_canvas1');
+  let stream1;
+  let vremote1;
+
+  const h2 = new CaptureStreamTestHelper2D(50, 50);
+  let canvas2;
+  let stream2;
+  let vremote2;
+
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function DRAW_INITIAL_LOCAL_GREEN(test) {
+      h1.drawColor(canvas1, h1.green);
+    },
+    function PC_LOCAL_CANVAS_CAPTURESTREAM(test) {
+      stream1 = canvas1.captureStream(0);
+      test.pcLocal.attachLocalStream(stream1);
+      let i = 0;
+      return setInterval(function() {
+        try {
+          info("draw " + i ? "green" : "red");
+          h1.drawColor(canvas1, i ? h1.green : h1.red);
+          i = 1 - i;
+          stream1.requestFrame();
+          if (stream2 != null) {
+            h2.drawColor(canvas2, i ? h2.green : h2.blue);
+            stream2.requestFrame();
+          }
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
+    }
+  ]);
+
+  test.chain.append([
+    function FIND_REMOTE_VIDEO() {
+      vremote1 = test.pcRemote.remoteMediaElements[0];
+      ok(!!vremote1, "Should have remote video element for pcRemote");
+    },
+    function WAIT_FOR_REMOTE_GREEN() {
+      return h1.pixelMustBecome(vremote1, h1.green, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become green",
+      });
+    },
+    function WAIT_FOR_REMOTE_RED() {
+      return h1.pixelMustBecome(vremote1, h1.red, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become red",
+      });
+    }
+  ]);
+
+  addRenegotiation(test.chain,
+    [
+      function PC_LOCAL_ADD_SECOND_STREAM(test) {
+        canvas2 = h2.createAndAppendElement('canvas', 'source_canvas2');
+        h2.drawColor(canvas2, h2.blue);
+        stream2 = canvas2.captureStream(0);
+
+        // can't use test.pcLocal.getAllUserMediaAndAddStreams([{video: true}]);
+        // because it doesn't let us substitute the capture stream
+        test.pcLocal.attachLocalStream(stream2);
+      }
+    ]
+  );
+
+  test.chain.append([
+    function FIND_REMOTE2_VIDEO() {
+      vremote2 = test.pcRemote.remoteMediaElements[1];
+      ok(!!vremote2, "Should have remote2 video element for pcRemote");
+    },
+    function WAIT_FOR_REMOTE2_BLUE() {
+      return h2.pixelMustBecome(vremote2, h2.blue, {
+        threshold: 128,
+        infoString: "pcRemote's remote2 should become blue",
+      });
+    },
+    function DRAW_NEW_LOCAL_GREEN(test) {
+      stream1.requestFrame();
+      h1.drawColor(canvas1, h1.green);
+    },
+    function WAIT_FOR_REMOTE1_GREEN() {
+      return h1.pixelMustBecome(vremote1, h1.green, {
+        threshold: 128,
+        infoString: "pcRemote's remote1 should become green",
+      });
+    }
+  ]);
+
+  await test.run();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_videoCodecs.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_videoCodecs.html
new file mode 100644
index 0000000000..7a245b5d8c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_videoCodecs.html
@@ -0,0 +1,142 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1395853",
+    title: "Verify video content over WebRTC for every video codec",
+  });
+
+  async function testVideoCodec(options = {}, codec) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], []);
+
+    let payloadType;
+    test.chain.insertBefore("PC_LOCAL_SET_LOCAL_DESCRIPTION", [
+      function PC_LOCAL_FILTER_OUT_CODECS() {
+        const otherCodec = codecs.find(c => c != codec);
+        const otherId = sdputils.findCodecId(test.originalOffer.sdp, otherCodec.name, otherCodec.offset);
+        const otherRtpmapMatcher = new RegExp(`a=rtpmap:${otherId}.*\\r\\n`, "gi");
+
+        const id = sdputils.findCodecId(test.originalOffer.sdp, codec.name, codec.offset);
+        payloadType = Number(id);
+        if (codec.offset) {
+          isnot(id, sdputils.findCodecId(test.originalOffer.sdp, codec.name, 0),
+            "Different offsets should return different payload types");
+        }
+        test.originalOffer.sdp =
+          sdputils.removeAllButPayloadType(test.originalOffer.sdp, id);
+
+        ok(!test.originalOffer.sdp.match(new RegExp(`m=.*UDP/TLS/RTP/SAVPF.* ${otherId}[^0-9]`, "gi")),
+          `Other codec ${otherId} should be removed after filtering`);
+        ok(test.originalOffer.sdp.match(new RegExp(`m=.*UDP/TLS/RTP/SAVPF.* ${id}[^0-9]`, "gi")),
+          `Tested codec ${id} should remain after filtering`);
+
+        // We only set it now, or the framework would remove non-H264 codecs
+        // for us.
+        options.h264 = codec.name == "H264";
+      },
+    ]);
+
+    test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+      [PC_LOCAL_TEST_LOCAL_STATS]);
+
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      [PC_REMOTE_TEST_REMOTE_STATS]);
+
+    test.chain.append([
+      async function PC_LOCAL_TEST_CODEC() {
+        const stats = await test.pcLocal._pc.getStats();
+        let codecCount = 0;
+        stats.forEach(stat => {
+          if (stat.type == "codec") {
+            is(codecCount++, 0, "expected only one encode codec stat");
+            is(stat.payloadType, payloadType, "payloadType as expected");
+            is(stat.mimeType, `video/${codec.name}`, "mimeType as expected");
+            is(stat.codecType, "encode", "codecType as expected");
+          }
+        });
+      },
+      async function PC_REMOTE_TEST_CODEC() {
+        const stats = await test.pcRemote._pc.getStats();
+        let codecCount = 0;
+        stats.forEach(stat => {
+          if (stat.type == "codec") {
+            is(codecCount++, 0, "expected only one decode codec stat");
+            is(stat.payloadType, payloadType, "payloadType as expected");
+            is(stat.mimeType, `video/${codec.name}`, "mimeType as expected");
+            is(stat.codecType, "decode", "codecType as expected");
+          }
+        });
+      },
+      async function CHECK_VIDEO_FLOW() {
+        try {
+          const h = new VideoStreamHelper();
+          await h.checkVideoPlaying(
+              test.pcRemote.remoteMediaElements[0],
+              10, 10, 128);
+          ok(true, `Got video flow for codec ${codec.name}, offset ${codec.offset}`);
+        } catch(e) {
+          ok(false, `No video flow for codec ${codec.name}, offset ${codec.offset}: ${e}`);
+        }
+      },
+    ]);
+
+    await test.run();
+  }
+
+  // We match the name against the sdp to figure out the payload type,
+  // so all other present codecs can be removed.
+  // Use `offset` when there are multiple instances of a codec expected in an sdp.
+  const codecs = [
+    { name: "VP8" },
+    { name: "VP9" },
+    { name: "H264" },
+    { name: "H264", offset: 1 },
+  ];
+
+  runNetworkTest(async (options) => {
+    // This test expects the video being captured will change color. Use fake
+    // video device as loopback does not currently change.
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    for (let codec of codecs) {
+      info(`Testing video for codec ${codec.name} offset ${codec.offset}`);
+      try {
+        let enc = SpecialPowers.getBoolPref('media.webrtc.platformencoder');
+        let dec = SpecialPowers.getBoolPref('media.navigator.mediadatadecoder_h264_enabled');
+        if (codec.name == "H264") {
+          await matchPlatformH264CodecPrefs();
+          if (codec.offset == 1) {
+            // Force fake GMP codec for H.264 mode 0 because not all platforms
+            // support slice size control. Re-enable it after
+            // a. SW encoder fallback support (bug 1726617), and
+            // b. returning valid bitstream from fake GMP encoder (bug 1509012).
+            await pushPrefs(
+              ['media.webrtc.platformencoder', false],
+              ['media.navigator.mediadatadecoder_h264_enabled', false],
+            );
+          }
+        }
+        await testVideoCodec(options, codec);
+        await pushPrefs(
+          ['media.webrtc.platformencoder', enc],
+          ['media.navigator.mediadatadecoder_h264_enabled', dec],
+        );
+      } catch(e) {
+        ok(false, `Error in test for codec ${codec.name}: ${e}\n${e.stack}`);
+      }
+      info(`Tested video for codec ${codec.name}`);
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_videoRenegotiationInactiveAnswer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_videoRenegotiationInactiveAnswer.html
new file mode 100644
index 0000000000..b77633493d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_videoRenegotiationInactiveAnswer.html
@@ -0,0 +1,95 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="sdpUtils.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1213773",
+    title: "Renegotiation: answerer uses a=inactive for video"
+  });
+
+  runNetworkTest(async (options) => {
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    const emitter = new VideoFrameEmitter();
+    const helper = new VideoStreamHelper();
+
+    const test = new PeerConnectionTest(options);
+
+    test.chain.replace("PC_LOCAL_GUM", [
+      function PC_LOCAL_CANVAS_CAPTURESTREAM(test) {
+        test.pcLocal.attachLocalStream(emitter.stream());
+        emitter.start();
+      }
+    ]);
+
+    var haveFirstUnmuteEvent;
+
+    test.chain.insertBefore("PC_REMOTE_SET_LOCAL_DESCRIPTION", [
+      function PC_REMOTE_SETUP_ONUNMUTE_1() {
+        haveFirstUnmuteEvent = haveEvent(test.pcRemote._pc.getReceivers()[0].track, "unmute");
+      }
+    ]);
+
+    test.chain.append([
+      function PC_REMOTE_CHECK_VIDEO_UNMUTED() {
+        return haveFirstUnmuteEvent;
+      },
+      function PC_REMOTE_WAIT_FOR_FRAMES() {
+        var vremote = test.pcRemote.remoteMediaElements[0];
+        ok(vremote, "Should have remote video element for pcRemote");
+        return addFinallyToPromise(helper.checkVideoPlaying(vremote))
+            .finally(() => emitter.stop());
+      }
+    ]);
+
+    addRenegotiation(test.chain, []);
+
+    test.chain.insertAfter("PC_LOCAL_GET_ANSWER", [
+        function PC_LOCAL_REWRITE_REMOTE_SDP_INACTIVE(test) {
+          test._remote_answer.sdp =
+            sdputils.setAllMsectionsInactive(test._remote_answer.sdp);
+        }
+    ], false, 1);
+
+    test.chain.append([
+      function PC_REMOTE_ENSURE_NO_FRAMES() {
+        var vremote = test.pcRemote.remoteMediaElements[0];
+        ok(vremote, "Should have remote video element for pcRemote");
+        emitter.start();
+        return addFinallyToPromise(helper.checkVideoPaused(vremote))
+            .finally(() => emitter.stop());
+      },
+    ]);
+
+    test.chain.remove("PC_REMOTE_CHECK_STATS", 1);
+    test.chain.remove("PC_LOCAL_CHECK_STATS", 1);
+
+    addRenegotiation(test.chain, []);
+
+    test.chain.append([
+      function PC_REMOTE_WAIT_FOR_FRAMES_2() {
+        var vremote = test.pcRemote.remoteMediaElements[0];
+        ok(vremote, "Should have remote video element for pcRemote");
+        emitter.start();
+        return addFinallyToPromise(helper.checkVideoPlaying(vremote))
+            .finally(() => emitter.stop());
+      }
+    ]);
+
+    test.setMediaConstraints([{video: true}], []);
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_webAudio.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_webAudio.html
new file mode 100644
index 0000000000..1d695ecbfa
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_webAudio.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1081819",
+  title: "WebAudio on both input and output side of peerconnection"
+});
+
+// This tests WebAudio (a 700Hz OscillatorNode) as input to a PeerConnection.
+// It also tests that a PeerConnection works as input to WebAudio as the remote
+// stream is connected to an AnalyserNode and compared to the source node.
+
+runNetworkTest(function() {
+  const test = new PeerConnectionTest();
+  test.audioContext = new AudioContext();
+  test.setMediaConstraints([{audio: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function PC_LOCAL_WEBAUDIO_SOURCE(test) {
+      const oscillator = test.audioContext.createOscillator();
+      oscillator.type = 'sine';
+      oscillator.frequency.value = 700;
+      oscillator.start();
+      const dest = test.audioContext.createMediaStreamDestination();
+      oscillator.connect(dest);
+      test.pcLocal.attachLocalStream(dest.stream);
+    }
+  ]);
+  test.chain.append([
+    function CHECK_AUDIO_FLOW(test) {
+      return test.pcRemote.checkReceivingToneFrom(test.audioContext, test.pcLocal);
+    }
+  ]);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_selftest.html b/dom/media/webrtc/tests/mochitests/test_selftest.html
new file mode 100644
index 0000000000..3f1ce1402d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_selftest.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "Self-test of harness functions",
+    visible: true
+  });
+
+function TEST(test) {}
+
+var catcher = func => {
+  try {
+    func();
+    return null;
+  } catch (e) {
+    return e.message;
+  }
+};
+
+runNetworkTest(() => {
+  var test = new PeerConnectionTest();
+  test.setMediaConstraints([{video: true}], [{video: true}]);
+  is(catcher(() => test.chain.replace("PC_LOCAL_SET_LOCAL_DESCRIPTION", TEST)),
+     null, "test.chain.replace works");
+  is(catcher(() => test.chain.replace("FOO", TEST)),
+     "Unknown test: FOO", "test.chain.replace catches typos");
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_setSinkId.html b/dom/media/webrtc/tests/mochitests/test_setSinkId.html
new file mode 100644
index 0000000000..0d85114a0e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_setSinkId.html
@@ -0,0 +1,83 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+
+<script>
+  createHTML({
+    title: "SetSinkId in HTMLMediaElement",
+    bug: "934425",
+  });
+
+  const memoryReportPath = 'explicit/media/media-manager-aggregates';
+
+  /**
+   * Run a test to verify set sink id in audio element.
+   */
+  runTest(async () => {
+    await pushPrefs(["media.setsinkid.enabled", true]);
+
+    if (!SpecialPowers.getCharPref("media.audio_loopback_dev", "")) {
+      ok(false, "No loopback device set by framework. Try --use-test-media-devices");
+      return;
+    }
+
+    // Expose an audio output device.
+    SpecialPowers.wrap(document).notifyUserGestureActivation();
+    await navigator.mediaDevices.selectAudioOutput();
+
+    const allDevices = await navigator.mediaDevices.enumerateDevices();
+    const audioDevices = allDevices.filter(({kind}) => kind == 'audiooutput');
+    is(audioDevices.length, 1, "Number of output devices found");
+
+    const audio = createMediaElement("audio", "audio");
+    document.body.appendChild(audio);
+
+    is(audio.sinkId, "", "Initial value is empty string");
+
+    const p = audio.setSinkId(audioDevices[0].deviceId);
+    is(audio.sinkId, "", "Value is unchanged upon function return");
+    is(await p, undefined, "promise resolves with undefined");
+    is(audio.sinkId, audioDevices[0].deviceId, `Sink device is set, id: ${audio.sinkId}`);
+
+    await audio.setSinkId(audioDevices[0].deviceId);
+    ok(true, `Sink device is set for 2nd time for the same id: ${audio.sinkId}`);
+
+    try {
+      await audio.setSinkId("dummy sink id");
+      ok(false, "Never enter here, this must fail");
+    } catch (error) {
+      ok(true, `Set sink id expected to fail: ${error}`);
+      is(error.name, "NotFoundError", "Verify correct error");
+    }
+
+    const {usage: usage1} =
+          await collectMemoryUsage(memoryReportPath); // Provided by head.js
+
+    ok(usage1 > 0, "MediaManager memory usage should be non-zero to store \
+device ids after enumerateDevices");
+
+    const p2 = audio.setSinkId("");
+    is(audio.sinkId, audioDevices[0].deviceId,
+       'sinkId after setSinkId("") return');
+    is(await p2, undefined,
+       "promise resolution value when sinkId parameter is empty");
+    is(audio.sinkId, "", 'sinkId after setSinkId("") resolution');
+
+    await audio.setSinkId(audioDevices[0].deviceId);
+
+    const {usage: usage2, reportCount} =
+          await collectMemoryUsage(memoryReportPath);
+    is(reportCount, 1,
+       'Expect only one MediaManager to report in content processes.');
+    is(usage2, usage1, "MediaManager memory usage should return to previous \
+value after promise resolution");
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_setSinkId_default_addTrack.html b/dom/media/webrtc/tests/mochitests/test_setSinkId_default_addTrack.html
new file mode 100644
index 0000000000..64db4cad7c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_setSinkId_default_addTrack.html
@@ -0,0 +1,52 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+
+<script>
+  createHTML({
+    title: "HTMLMediaElement.setSinkId with default device and adding a track",
+    bug: "1661649",
+  });
+
+  /**
+   * Run a test to verify set sink id in audio element.
+   */
+  runTest(async () => {
+    await pushPrefs(["media.setsinkid.enabled", true]);
+
+    // Expose an audio output device.
+    SpecialPowers.wrap(document).notifyUserGestureActivation();
+    await navigator.mediaDevices.selectAudioOutput();
+
+    const allDevices = await navigator.mediaDevices.enumerateDevices();
+    const audioDevices = allDevices.filter(({kind}) => kind == 'audiooutput');
+    info(`Found  ${audioDevices.length} output devices`);
+    isnot(audioDevices.length, 0, "Found output devices");
+
+    const audio = createMediaElement("audio", "audio");
+    document.body.appendChild(audio);
+
+    audio.srcObject = await navigator.mediaDevices.getUserMedia({audio: true});
+    audio.play();
+
+    await audio.setSinkId(audioDevices[0].deviceId);
+    await audio.setSinkId("");
+    is(audio.sinkId, "", "sinkId restored to default");
+
+    audio.srcObject.addTrack((await navigator.mediaDevices.getUserMedia({audio: true})).getTracks()[0]);
+
+    await wait(0);
+
+    for (let t of audio.srcObject.getTracks()) {
+      t.stop();
+    }
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_setSinkId_preMutedElement.html b/dom/media/webrtc/tests/mochitests/test_setSinkId_preMutedElement.html
new file mode 100644
index 0000000000..fb65c3312f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_setSinkId_preMutedElement.html
@@ -0,0 +1,100 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "Test changing sink and muting before the MediaStream is set",
+  bug: "1651049",
+  visible: true
+});
+
+let getOutputDeviceId = async () => {
+  let devices = await navigator.mediaDevices.enumerateDevices();
+  let audios = devices.filter(d => d.kind == "audiooutput");
+  ok(audios.length, "One or more output devices found.");
+  return audios[0].deviceId;
+}
+
+let verifyAudioTone = async (ac, stream, freq) => {
+  const toneAnalyser = new AudioStreamAnalyser(ac, stream);
+  return toneAnalyser.waitForAnalysisSuccess(array => {
+    const lowerFreq = freq / 2;
+    const upperFreq = freq + 1000;
+    const lowerMag = array[toneAnalyser.binIndexForFrequency(lowerFreq)];
+    const freqMag = array[toneAnalyser.binIndexForFrequency(freq)];
+    const upperMag = array[toneAnalyser.binIndexForFrequency(upperFreq)];
+    info("Audio tone expected. "
+         + lowerFreq + ": " + lowerMag + ", "
+         + freq + ": " + freqMag + ", "
+         + upperFreq + ": " + upperMag);
+    return lowerMag < 50 && freqMag > 200 && upperMag < 50;
+  });
+}
+
+let verifyNoAudioTone = async (ac, stream, freq) => {
+  const toneAnalyser = new AudioStreamAnalyser(ac, stream);
+  // repeat check 100 times to make sure that it is muted.
+  let retryCnt = 0;
+  return toneAnalyser.waitForAnalysisSuccess(array => {
+    const lowerFreq = freq / 2;
+    const upperFreq = freq + 1000;
+    const lowerMag = array[toneAnalyser.binIndexForFrequency(lowerFreq)];
+    const freqMag = array[toneAnalyser.binIndexForFrequency(freq)];
+    const upperMag = array[toneAnalyser.binIndexForFrequency(upperFreq)];
+    info("No audio tone expected. "
+         + lowerFreq + ": " + lowerMag + ", "
+         + freq + ": " + freqMag + ", "
+         + upperFreq + ": " + upperMag);
+    return lowerMag == 0 && freqMag == 0 && upperMag == 0 && ++retryCnt == 100;
+  });
+}
+
+runTest(async () => {
+  let audioDevice = SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (!audioDevice) {
+    todo(false, "No loopback device set by framework. Try --use-test-media-devices");
+    return;
+  }
+
+  await pushPrefs(["media.setsinkid.enabled", true]);
+
+  // Implicitly expose the loopback device sink by opening the source in the
+  // same group.
+  const verifyStream = await getUserMedia({audio: true});
+  // We gonna test our tone, stop the auto created one.
+  DefaultLoopbackTone.stop();
+
+  let sinkId = await getOutputDeviceId();
+  isnot(sinkId, "", "SinkId is not null");
+
+  let audioElement = createMediaElement('audio', 'audioElement');
+  audioElement.muted = true;
+  await audioElement.setSinkId(sinkId);
+  isnot(audioElement.sinkId, "", "sinkId property of the element is not null");
+
+  // The test stream is a sine tone of 1000 Hz
+  let ac = new AudioContext();
+  const frequency = 2000;
+  let stream = createOscillatorStream(ac, frequency);
+  await verifyAudioTone(ac, stream, frequency);
+
+  audioElement.srcObject = stream;
+  audioElement.play();
+
+  // Verify the silent output using the loopback device.
+  await verifyNoAudioTone(ac, verifyStream, frequency);
+  info("output is muted");
+
+  // Clean up
+  audioElement.pause();
+  audioElement.srcObject = null;
+  verifyStream.getTracks()[0].stop();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_unfocused_pref.html b/dom/media/webrtc/tests/mochitests/test_unfocused_pref.html
new file mode 100644
index 0000000000..22df020f7c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_unfocused_pref.html
@@ -0,0 +1,49 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <meta charset="utf-8">
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<script>
+"use strict";
+
+createHTML({
+  // This pref exists only for a partner testing framework without WebDriver
+  // switch-to-window nor SpecialPowers to set the active window.
+  // Prefer "focusmanager.testmode".
+  title: "Test media.devices.unfocused.enabled",
+  bug: "1740824"
+});
+
+const blank_url = "/tests/docshell/test/navigation/blank.html";
+
+async function resolveOnEvent(target, name) {
+  return new Promise(r => target.addEventListener(name, r, {once: true}));
+}
+
+runTest(async () => {
+  ok(document.hasFocus(), "This test expects initial focus on the document.");
+  // 'resizable' is requested for a separate OS window on relevant platforms
+  // so that this test tests OS focus changes rather than document visibility.
+  const other = window.open(blank_url, "", "resizable");
+  SimpleTest.registerCleanupFunction(() => {
+    other.close();
+    return SimpleTest.promiseFocus(window);
+  });
+  await Promise.all([
+    resolveOnEvent(window, 'blur'),
+    SimpleTest.promiseFocus(other),
+    pushPrefs(["media.devices.unfocused.enabled", true]),
+  ]);
+  ok(!document.hasFocus(), "!document.hasFocus()");
+  await navigator.mediaDevices.enumerateDevices();
+  ok(true, "enumerateDevices() completes without focus.");
+  // The focus requirement with media.devices.unfocused.enabled false
+  // (default) is tested in
+  // testing/web-platform/mozilla/tests/mediacapture-streams/enumerateDevices-without-focus.https.html
+});
+
+</script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/turnConfig.js b/dom/media/webrtc/tests/mochitests/turnConfig.js
new file mode 100644
index 0000000000..1267de4ec5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/turnConfig.js
@@ -0,0 +1,16 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/* An example of how to specify two TURN server configs:
+ *
+ * Note: If turn URL uses FQDN rather then an IP address the TURN relay
+ *       verification step in checkStatsIceConnectionType might fail.
+ *
+ * var turnServers = {
+ *   local: { iceServers: [{"username":"mozilla","credential":"mozilla","url":"turn:10.0.0.1"}] },
+ *   remote: { iceServers: [{"username":"firefox","credential":"firefox","url":"turn:10.0.0.2"}] }
+ * };
+ */
+
+var turnServers = {};
author	Daniel Baumann <daniel.baumann@progress-linux.org>	2024-04-07 19:33:14 +0000
committer	Daniel Baumann <daniel.baumann@progress-linux.org>	2024-04-07 19:33:14 +0000
commit	36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree	105e8c98ddea1c1e4784a60a5a6410fa416be2de /dom/media/webrtc/tests
parent	Initial commit. (diff)
download	firefox-esr-upstream.tar.xz firefox-esr-upstream.zip