Adding upstream version 115.7.0esr.upstream/115.7.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

264 files changed, 30134 insertions, 0 deletions

diff --git a/third_party/libwebrtc/examples/BUILD.gn b/third_party/libwebrtc/examples/BUILD.gn
new file mode 100644
index 0000000000..18cb6ecc67
--- /dev/null
+++ b/third_party/libwebrtc/examples/BUILD.gn
@@ -0,0 +1,953 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../webrtc.gni")
+
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+} else if (is_mac) {
+  import("//build/config/mac/rules.gni")
+} else if (is_ios) {
+  import("//build/config/ios/rules.gni")
+} else if (is_linux || is_chromeos) {
+  import("//build/config/linux/pkg_config.gni")
+}
+
+group("examples") {
+  # This target shall build all targets in examples.
+  testonly = true
+  deps = []
+
+  if (is_android) {
+    deps += [
+      ":AppRTCMobile",
+      ":AppRTCMobile_test_apk",
+      ":libwebrtc_unity",
+      "androidvoip",
+    ]
+
+    # TODO(sakal): We include some code from the tests. Remove this dependency
+    # and remove this if-clause.
+    if (rtc_include_tests) {
+      deps += [ "androidnativeapi" ]
+    }
+  }
+
+  if (!build_with_chromium) {
+    deps += [ ":stun_prober" ]
+  }
+
+  if (is_ios || (is_mac && target_cpu != "x86")) {
+    deps += [ ":AppRTCMobile" ]
+  }
+
+  if (is_linux || is_chromeos || is_win) {
+    deps += [
+      ":peerconnection_server",
+      ":stunserver",
+      ":turnserver",
+    ]
+    if (current_os != "winuwp") {
+      deps += [ ":peerconnection_client" ]
+    }
+  }
+
+  if (is_android || is_win) {
+    deps += [ ":webrtc_unity_plugin" ]
+  }
+}
+
+rtc_library("read_auth_file") {
+  testonly = true
+  sources = [
+    "turnserver/read_auth_file.cc",
+    "turnserver/read_auth_file.h",
+  ]
+  deps = [
+    "../api:array_view",
+    "../rtc_base:stringutils",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ]
+}
+
+if (rtc_include_tests) {
+  rtc_test("examples_unittests") {
+    testonly = true
+    sources = [ "turnserver/read_auth_file_unittest.cc" ]
+    deps = [
+      ":read_auth_file",
+      "../test:test_main",
+      "//test:test_support",
+      "//testing/gtest",
+    ]
+  }
+}
+
+if (is_android) {
+  rtc_android_apk("AppRTCMobile") {
+    testonly = true
+    apk_name = "AppRTCMobile"
+    android_manifest = "androidapp/AndroidManifest.xml"
+    min_sdk_version = 21
+    target_sdk_version = 31
+
+    deps = [
+      ":AppRTCMobile_javalib",
+      ":AppRTCMobile_resources",
+      "../rtc_base:base_java",
+    ]
+
+    shared_libraries = [ "../sdk/android:libjingle_peerconnection_so" ]
+  }
+
+  rtc_android_library("AppRTCMobile_javalib") {
+    testonly = true
+    android_manifest = "androidapp/AndroidManifest.xml"
+
+    sources = [
+      "androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java",
+      "androidapp/src/org/appspot/apprtc/AppRTCBluetoothManager.java",
+      "androidapp/src/org/appspot/apprtc/AppRTCClient.java",
+      "androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java",
+      "androidapp/src/org/appspot/apprtc/CallActivity.java",
+      "androidapp/src/org/appspot/apprtc/CallFragment.java",
+      "androidapp/src/org/appspot/apprtc/CaptureQualityController.java",
+      "androidapp/src/org/appspot/apprtc/ConnectActivity.java",
+      "androidapp/src/org/appspot/apprtc/CpuMonitor.java",
+      "androidapp/src/org/appspot/apprtc/DirectRTCClient.java",
+      "androidapp/src/org/appspot/apprtc/HudFragment.java",
+      "androidapp/src/org/appspot/apprtc/PeerConnectionClient.java",
+      "androidapp/src/org/appspot/apprtc/RecordedAudioToFileController.java",
+      "androidapp/src/org/appspot/apprtc/RoomParametersFetcher.java",
+      "androidapp/src/org/appspot/apprtc/RtcEventLog.java",
+      "androidapp/src/org/appspot/apprtc/SettingsActivity.java",
+      "androidapp/src/org/appspot/apprtc/SettingsFragment.java",
+      "androidapp/src/org/appspot/apprtc/TCPChannelClient.java",
+      "androidapp/src/org/appspot/apprtc/UnhandledExceptionHandler.java",
+      "androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java",
+      "androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java",
+      "androidapp/src/org/appspot/apprtc/util/AppRTCUtils.java",
+      "androidapp/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java",
+    ]
+
+    resources_package = "org.appspot.apprtc"
+    deps = [
+      ":AppRTCMobile_resources",
+      "../rtc_base:base_java",
+      "../sdk/android:audio_api_java",
+      "../sdk/android:base_java",
+      "../sdk/android:camera_java",
+      "../sdk/android:default_video_codec_factory_java",
+      "../sdk/android:filevideo_java",
+      "../sdk/android:hwcodecs_java",
+      "../sdk/android:java_audio_device_module_java",
+      "../sdk/android:libjingle_peerconnection_java",
+      "../sdk/android:libjingle_peerconnection_metrics_default_java",
+      "../sdk/android:peerconnection_java",
+      "../sdk/android:screencapturer_java",
+      "../sdk/android:surfaceviewrenderer_java",
+      "../sdk/android:swcodecs_java",
+      "../sdk/android:video_api_java",
+      "../sdk/android:video_java",
+      "androidapp/third_party/autobanh:autobanh_java",
+      "//third_party/androidx:androidx_annotation_annotation_java",
+    ]
+  }
+
+  android_resources("AppRTCMobile_resources") {
+    testonly = true
+    sources = [
+      "androidapp/res/drawable-hdpi/disconnect.png",
+      "androidapp/res/drawable-hdpi/ic_action_full_screen.png",
+      "androidapp/res/drawable-hdpi/ic_action_return_from_full_screen.png",
+      "androidapp/res/drawable-hdpi/ic_launcher.png",
+      "androidapp/res/drawable-hdpi/ic_loopback_call.png",
+      "androidapp/res/drawable-ldpi/disconnect.png",
+      "androidapp/res/drawable-ldpi/ic_action_full_screen.png",
+      "androidapp/res/drawable-ldpi/ic_action_return_from_full_screen.png",
+      "androidapp/res/drawable-ldpi/ic_launcher.png",
+      "androidapp/res/drawable-ldpi/ic_loopback_call.png",
+      "androidapp/res/drawable-mdpi/disconnect.png",
+      "androidapp/res/drawable-mdpi/ic_action_full_screen.png",
+      "androidapp/res/drawable-mdpi/ic_action_return_from_full_screen.png",
+      "androidapp/res/drawable-mdpi/ic_launcher.png",
+      "androidapp/res/drawable-mdpi/ic_loopback_call.png",
+      "androidapp/res/drawable-xhdpi/disconnect.png",
+      "androidapp/res/drawable-xhdpi/ic_action_full_screen.png",
+      "androidapp/res/drawable-xhdpi/ic_action_return_from_full_screen.png",
+      "androidapp/res/drawable-xhdpi/ic_launcher.png",
+      "androidapp/res/drawable-xhdpi/ic_loopback_call.png",
+      "androidapp/res/layout/activity_call.xml",
+      "androidapp/res/layout/activity_connect.xml",
+      "androidapp/res/layout/fragment_call.xml",
+      "androidapp/res/layout/fragment_hud.xml",
+      "androidapp/res/menu/connect_menu.xml",
+      "androidapp/res/values-v17/styles.xml",
+      "androidapp/res/values-v21/styles.xml",
+      "androidapp/res/values/arrays.xml",
+      "androidapp/res/values/strings.xml",
+      "androidapp/res/xml/preferences.xml",
+    ]
+
+    # Needed for Bazel converter.
+    custom_package = "org.appspot.apprtc"
+    resource_dirs = [ "androidapp/res" ]
+    assert(resource_dirs != [])  # Mark as used.
+  }
+
+  rtc_instrumentation_test_apk("AppRTCMobile_test_apk") {
+    apk_name = "AppRTCMobileTest"
+    android_manifest = "androidtests/AndroidManifest.xml"
+    min_sdk_version = 21
+    target_sdk_version = 31
+
+    sources = [
+      "androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java",
+    ]
+
+    apk_under_test = ":AppRTCMobile"
+
+    deps = [
+      ":AppRTCMobile_javalib",
+      "../sdk/android:base_java",
+      "../sdk/android:camera_java",
+      "../sdk/android:libjingle_peerconnection_java",
+      "../sdk/android:peerconnection_java",
+      "../sdk/android:video_api_java",
+      "../sdk/android:video_java",
+      "//third_party/android_support_test_runner:runner_java",
+      "//third_party/androidx:androidx_test_runner_java",
+      "//third_party/junit",
+    ]
+  }
+}
+
+if (is_ios || (is_mac && target_cpu != "x86")) {
+  config("apprtc_common_config") {
+    include_dirs = [ "objc/AppRTCMobile/common" ]
+  }
+
+  rtc_library("apprtc_common") {
+    testonly = true
+    sources = [
+      "objc/AppRTCMobile/common/ARDUtilities.h",
+      "objc/AppRTCMobile/common/ARDUtilities.m",
+    ]
+    public_configs = [ ":apprtc_common_config" ]
+    deps = [ "../sdk:base_objc" ]
+  }
+
+  config("apprtc_signaling_config") {
+    include_dirs = [ "objc/AppRTCMobile" ]
+  }
+
+  rtc_library("apprtc_signaling") {
+    testonly = true
+    sources = [
+      "objc/AppRTCMobile/ARDAppClient+Internal.h",
+      "objc/AppRTCMobile/ARDAppClient.h",
+      "objc/AppRTCMobile/ARDAppClient.m",
+      "objc/AppRTCMobile/ARDAppEngineClient.h",
+      "objc/AppRTCMobile/ARDAppEngineClient.m",
+      "objc/AppRTCMobile/ARDCaptureController.h",
+      "objc/AppRTCMobile/ARDCaptureController.m",
+      "objc/AppRTCMobile/ARDExternalSampleCapturer.h",
+      "objc/AppRTCMobile/ARDExternalSampleCapturer.m",
+      "objc/AppRTCMobile/ARDJoinResponse+Internal.h",
+      "objc/AppRTCMobile/ARDJoinResponse.h",
+      "objc/AppRTCMobile/ARDJoinResponse.m",
+      "objc/AppRTCMobile/ARDMessageResponse+Internal.h",
+      "objc/AppRTCMobile/ARDMessageResponse.h",
+      "objc/AppRTCMobile/ARDMessageResponse.m",
+      "objc/AppRTCMobile/ARDRoomServerClient.h",
+      "objc/AppRTCMobile/ARDSettingsModel+Private.h",
+      "objc/AppRTCMobile/ARDSettingsModel.h",
+      "objc/AppRTCMobile/ARDSettingsModel.m",
+      "objc/AppRTCMobile/ARDSettingsStore.h",
+      "objc/AppRTCMobile/ARDSettingsStore.m",
+      "objc/AppRTCMobile/ARDSignalingChannel.h",
+      "objc/AppRTCMobile/ARDSignalingMessage.h",
+      "objc/AppRTCMobile/ARDSignalingMessage.m",
+      "objc/AppRTCMobile/ARDStatsBuilder.h",
+      "objc/AppRTCMobile/ARDStatsBuilder.m",
+      "objc/AppRTCMobile/ARDTURNClient+Internal.h",
+      "objc/AppRTCMobile/ARDTURNClient.h",
+      "objc/AppRTCMobile/ARDTURNClient.m",
+      "objc/AppRTCMobile/ARDWebSocketChannel.h",
+      "objc/AppRTCMobile/ARDWebSocketChannel.m",
+      "objc/AppRTCMobile/RTCIceCandidate+JSON.h",
+      "objc/AppRTCMobile/RTCIceCandidate+JSON.m",
+      "objc/AppRTCMobile/RTCIceServer+JSON.h",
+      "objc/AppRTCMobile/RTCIceServer+JSON.m",
+      "objc/AppRTCMobile/RTCSessionDescription+JSON.h",
+      "objc/AppRTCMobile/RTCSessionDescription+JSON.m",
+    ]
+    public_configs = [ ":apprtc_signaling_config" ]
+    deps = [
+      ":apprtc_common",
+      ":socketrocket",
+      "../sdk:base_objc",
+      "../sdk:default_codec_factory_objc",
+      "../sdk:file_logger_objc",
+      "../sdk:helpers_objc",
+      "../sdk:mediaconstraints_objc",
+      "../sdk:peerconnectionfactory_base_objc",
+      "../sdk:videocapture_objc",
+      "../sdk:videoframebuffer_objc",
+      "../sdk:videosource_objc",
+    ]
+    frameworks = [
+      "CoreMedia.framework",
+      "QuartzCore.framework",
+    ]
+  }
+
+  if (is_ios) {
+    rtc_library("AppRTCMobile_lib") {
+      # iOS must use WebRTC.framework which is dynamically linked.
+      testonly = true
+      sources = [
+        "objc/AppRTCMobile/ios/ARDAppDelegate.h",
+        "objc/AppRTCMobile/ios/ARDAppDelegate.m",
+        "objc/AppRTCMobile/ios/ARDFileCaptureController.h",
+        "objc/AppRTCMobile/ios/ARDFileCaptureController.m",
+        "objc/AppRTCMobile/ios/ARDMainView.h",
+        "objc/AppRTCMobile/ios/ARDMainView.m",
+        "objc/AppRTCMobile/ios/ARDMainViewController.h",
+        "objc/AppRTCMobile/ios/ARDMainViewController.m",
+        "objc/AppRTCMobile/ios/ARDSettingsViewController.h",
+        "objc/AppRTCMobile/ios/ARDSettingsViewController.m",
+        "objc/AppRTCMobile/ios/ARDStatsView.h",
+        "objc/AppRTCMobile/ios/ARDStatsView.m",
+        "objc/AppRTCMobile/ios/ARDVideoCallView.h",
+        "objc/AppRTCMobile/ios/ARDVideoCallView.m",
+        "objc/AppRTCMobile/ios/ARDVideoCallViewController.h",
+        "objc/AppRTCMobile/ios/ARDVideoCallViewController.m",
+        "objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.h",
+        "objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.m",
+        "objc/AppRTCMobile/ios/UIImage+ARDUtilities.h",
+        "objc/AppRTCMobile/ios/UIImage+ARDUtilities.m",
+      ]
+
+      configs += [ "..:common_objc" ]
+
+      deps = [
+        ":apprtc_common",
+        ":apprtc_signaling",
+        "../sdk:audio_session_objc",
+        "../sdk:base_objc",
+        "../sdk:helpers_objc",
+        "../sdk:mediaconstraints_objc",
+        "../sdk:metal_objc",
+        "../sdk:peerconnectionfactory_base_objc",
+        "../sdk:peerconnectionfactory_base_objc",
+        "../sdk:videocapture_objc",
+        "../sdk:videocodec_objc",
+      ]
+      if (rtc_ios_macos_use_opengl_rendering) {
+        deps += [ "../sdk:opengl_ui_objc" ]
+      }
+
+      frameworks = [ "AVFoundation.framework" ]
+    }
+
+    ios_app_bundle("AppRTCMobile") {
+      testonly = true
+      sources = [ "objc/AppRTCMobile/ios/main.m" ]
+
+      info_plist = "objc/AppRTCMobile/ios/Info.plist"
+
+      configs += [ "..:common_config" ]
+      public_configs = [ "..:common_inherited_config" ]
+
+      deps = [
+        ":AppRTCMobile_ios_bundle_data",
+        ":AppRTCMobile_lib",
+        "../sdk:framework_objc",
+        "../sdk:ios_framework_bundle",
+      ]
+
+      if (rtc_apprtcmobile_broadcast_extension) {
+        deps += [
+          ":AppRTCMobileBroadcastSetupUI_extension_bundle",
+          ":AppRTCMobileBroadcastUpload_extension_bundle",
+        ]
+      }
+
+      if (target_cpu == "x86") {
+        deps += [ "//testing/iossim" ]
+      }
+    }
+
+    if (rtc_apprtcmobile_broadcast_extension) {
+      bundle_data("AppRTCMobileBroadcastUpload_extension_bundle") {
+        testonly = true
+        public_deps = [  # no-presubmit-check TODO(webrtc:8603)
+          ":AppRTCMobileBroadcastUpload",  # prevent code format
+        ]
+        sources = [ "$root_out_dir/AppRTCMobileBroadcastUpload.appex" ]
+        outputs = [ "{{bundle_contents_dir}}/Plugins/{{source_file_part}}" ]
+      }
+
+      bundle_data("AppRTCMobileBroadcastSetupUI_extension_bundle") {
+        testonly = true
+        public_deps = [  # no-presubmit-check TODO(webrtc:8603)
+          ":AppRTCMobileBroadcastSetupUI",  # prevent code format
+        ]
+        sources = [ "$root_out_dir/AppRTCMobileBroadcastSetupUI.appex" ]
+        outputs = [ "{{bundle_contents_dir}}/Plugins/{{source_file_part}}" ]
+      }
+
+      rtc_library("AppRTCMobileBroadcastUpload_lib") {
+        testonly = true
+        sources = [
+          "objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h",
+          "objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.m",
+        ]
+
+        deps = [
+          ":apprtc_signaling",
+          "../sdk:framework_objc+link",
+          "../sdk:ios_framework_bundle",
+        ]
+
+        frameworks = [ "ReplayKit.framework" ]
+      }
+
+      ios_appex_bundle("AppRTCMobileBroadcastUpload") {
+        testonly = true
+        configs += [ "..:common_config" ]
+        public_configs = [ "..:common_inherited_config" ]
+
+        info_plist = "objc/AppRTCMobile/ios/broadcast_extension/BroadcastUploadInfo.plist"
+
+        deps = [
+          ":AppRTCMobileBroadcastUpload_lib",
+          "../sdk:framework_objc",
+        ]
+      }
+
+      ios_appex_bundle("AppRTCMobileBroadcastSetupUI") {
+        sources = [
+          "objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.h",
+          "objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.m",
+        ]
+
+        info_plist = "objc/AppRTCMobile/ios/broadcast_extension/BroadcastSetupUIInfo.plist"
+
+        frameworks = [ "ReplayKit.framework" ]
+
+        deps = [ ":AppRTCMobile_ios_bundle_data" ]
+      }
+    }
+
+    bundle_data("AppRTCMobile_ios_bundle_data") {
+      sources = [
+        "objc/AppRTCMobile/ios/resources/Roboto-Regular.ttf",
+
+        # Sample video taken from https://media.xiph.org/video/derf/
+        "objc/AppRTCMobile/ios/resources/foreman.mp4",
+        "objc/AppRTCMobile/ios/resources/iPhone5@2x.png",
+        "objc/AppRTCMobile/ios/resources/iPhone6@2x.png",
+        "objc/AppRTCMobile/ios/resources/iPhone6p@3x.png",
+        "objc/AppRTCMobile/ios/resources/ic_call_end_black_24dp.png",
+        "objc/AppRTCMobile/ios/resources/ic_call_end_black_24dp@2x.png",
+        "objc/AppRTCMobile/ios/resources/ic_clear_black_24dp.png",
+        "objc/AppRTCMobile/ios/resources/ic_clear_black_24dp@2x.png",
+        "objc/AppRTCMobile/ios/resources/ic_settings_black_24dp.png",
+        "objc/AppRTCMobile/ios/resources/ic_settings_black_24dp@2x.png",
+        "objc/AppRTCMobile/ios/resources/ic_surround_sound_black_24dp.png",
+        "objc/AppRTCMobile/ios/resources/ic_surround_sound_black_24dp@2x.png",
+        "objc/AppRTCMobile/ios/resources/ic_switch_video_black_24dp.png",
+        "objc/AppRTCMobile/ios/resources/ic_switch_video_black_24dp@2x.png",
+        "objc/AppRTCMobile/ios/resources/mozart.mp3",
+        "objc/Icon-120.png",
+        "objc/Icon-180.png",
+        "objc/Icon.png",
+      ]
+      outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+    }
+
+    rtc_library("ObjCNativeAPIDemo_lib") {
+      testonly = true
+      sources = [
+        "objcnativeapi/objc/NADAppDelegate.h",
+        "objcnativeapi/objc/NADAppDelegate.m",
+        "objcnativeapi/objc/NADViewController.h",
+        "objcnativeapi/objc/NADViewController.mm",
+        "objcnativeapi/objc/objc_call_client.h",
+        "objcnativeapi/objc/objc_call_client.mm",
+      ]
+
+      deps = [
+        "../api:libjingle_peerconnection_api",
+        "../api:scoped_refptr",
+        "../api:sequence_checker",
+        "../api/audio_codecs:builtin_audio_decoder_factory",
+        "../api/audio_codecs:builtin_audio_encoder_factory",
+        "../api/rtc_event_log:rtc_event_log_factory",
+        "../api/task_queue:default_task_queue_factory",
+        "../media:rtc_audio_video",
+        "../modules/audio_processing",
+        "../modules/audio_processing:api",
+        "../pc:libjingle_peerconnection",
+        "../rtc_base/synchronization:mutex",
+        "../sdk:base_objc",
+        "../sdk:default_codec_factory_objc",
+        "../sdk:helpers_objc",
+        "../sdk:metal_objc",
+        "../sdk:native_api",
+        "../sdk:videocapture_objc",
+        "../sdk:videotoolbox_objc",
+      ]
+
+      if (rtc_ios_macos_use_opengl_rendering) {
+        deps += [ "../sdk:opengl_ui_objc" ]
+      }
+    }
+
+    ios_app_bundle("ObjCNativeAPIDemo") {
+      testonly = true
+      sources = [ "objcnativeapi/objc/main.m" ]
+
+      info_plist = "objcnativeapi/Info.plist"
+
+      configs += [ "..:common_config" ]
+      public_configs = [ "..:common_inherited_config" ]
+
+      deps = [ ":ObjCNativeAPIDemo_lib" ]
+
+      if (target_cpu == "x86") {
+        deps += [ "//testing/iossim" ]
+      }
+    }
+  }
+
+  if (is_mac) {
+    rtc_library("AppRTCMobile_lib") {
+      testonly = true
+      sources = [
+        "objc/AppRTCMobile/mac/APPRTCAppDelegate.h",
+        "objc/AppRTCMobile/mac/APPRTCAppDelegate.m",
+        "objc/AppRTCMobile/mac/APPRTCViewController.h",
+        "objc/AppRTCMobile/mac/APPRTCViewController.m",
+      ]
+      configs += [ "..:common_objc" ]
+      deps = [
+        ":apprtc_common",
+        ":apprtc_signaling",
+        "../sdk:base_objc",
+        "../sdk:helpers_objc",
+        "../sdk:mediaconstraints_objc",
+        "../sdk:metal_objc",
+        "../sdk:opengl_ui_objc",
+        "../sdk:peerconnectionfactory_base_objc",
+        "../sdk:peerconnectionfactory_base_objc",
+        "../sdk:videocapture_objc",
+        "../sdk:videocodec_objc",
+      ]
+    }
+
+    mac_app_bundle("AppRTCMobile") {
+      testonly = true
+      output_name = "AppRTCMobile"
+
+      sources = [ "objc/AppRTCMobile/mac/main.m" ]
+
+      public_configs = [ "..:common_inherited_config" ]
+
+      info_plist = "objc/AppRTCMobile/mac/Info.plist"
+
+      frameworks = [ "AppKit.framework" ]
+
+      ldflags = [
+        "-rpath",
+        "@executable_path/../Frameworks",
+      ]
+
+      deps = [
+        ":AppRTCMobile_lib",
+        "../sdk:mac_framework_bundle",
+        "../sdk:mac_framework_objc+link",
+      ]
+    }
+  }
+
+  config("socketrocket_include_config") {
+    include_dirs = [ "objc/AppRTCMobile/third_party/SocketRocket" ]
+  }
+
+  config("socketrocket_warning_config") {
+    # GN orders flags on a target before flags from configs. The default config
+    # adds these flags so to cancel them out they need to come from a config and
+    # cannot be on the target directly.
+    cflags = [
+      "-Wno-deprecated-declarations",
+      "-Wno-nonnull",
+      "-Wno-semicolon-before-method-body",
+      "-Wno-unused-variable",
+    ]
+
+    cflags_objc = [
+      # Enabled for cflags_objc in build/config/compiler/BUILD.gn.
+      "-Wno-objc-missing-property-synthesis",
+    ]
+  }
+
+  rtc_library("socketrocket") {
+    testonly = true
+    sources = [
+      "objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.h",
+      "objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m",
+    ]
+    configs += [ ":socketrocket_warning_config" ]
+    public_configs = [ ":socketrocket_include_config" ]
+
+    libs = [ "icucore" ]
+    frameworks = [
+      "CFNetwork.framework",
+      "Security.framework",
+    ]
+  }
+
+  if (rtc_include_tests) {
+    # TODO(kthelgason): compile xctests on mac when chromium supports it.
+    if (is_ios) {
+      rtc_library("apprtcmobile_test_sources") {
+        # iOS must use WebRTC.framework which is dynamically linked.
+        testonly = true
+        include_dirs = [
+          "objc/AppRTCMobile",
+          "objc/AppRTCMobile/ios",
+        ]
+        sources = [
+          "objc/AppRTCMobile/tests/ARDAppClient_xctest.mm",
+          "objc/AppRTCMobile/tests/ARDFileCaptureController_xctest.mm",
+          "objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm",
+        ]
+        deps = [
+          ":AppRTCMobile_lib",
+          ":apprtc_signaling",
+          "../rtc_base:ssl",
+          "../sdk:mediaconstraints_objc",
+          "../sdk:peerconnectionfactory_base_objc",
+          "../sdk:videocapture_objc",
+          "//build/config/ios:xctest",
+          "//third_party/ocmock",
+        ]
+      }
+
+      rtc_test("apprtcmobile_tests") {
+        is_xctest = true
+        info_plist = "objc/AppRTCMobile/ios/Info.plist"
+        sources = [ "objc/AppRTCMobile/tests/main.mm" ]
+        deps = [
+          ":AppRTCMobile_lib",
+          ":apprtcmobile_test_sources",
+          "../sdk:framework_objc",
+          "//test:test_support",
+        ]
+        ldflags = [ "-all_load" ]
+      }
+    }
+  }
+}
+
+if (is_linux || is_chromeos || is_win) {
+  if (is_linux || is_chromeos) {
+    pkg_config("gtk_config") {
+      packages = [
+        # Gtk requires gmodule, but it does not list it as a dependency in some
+        # misconfigured systems.
+        "gmodule-2.0",
+        "gthread-2.0",
+        "gtk+-3.0",
+      ]
+    }
+  }
+
+  rtc_executable("peerconnection_client") {
+    testonly = true
+    sources = [
+      "peerconnection/client/conductor.cc",
+      "peerconnection/client/conductor.h",
+      "peerconnection/client/defaults.cc",
+      "peerconnection/client/defaults.h",
+      "peerconnection/client/peer_connection_client.cc",
+      "peerconnection/client/peer_connection_client.h",
+    ]
+
+    deps = [
+      "../api:audio_options_api",
+      "../api:create_peerconnection_factory",
+      "../api:libjingle_peerconnection_api",
+      "../api:media_stream_interface",
+      "../api:rtp_sender_interface",
+      "../api:scoped_refptr",
+      "../api/audio:audio_mixer_api",
+      "../api/audio_codecs:audio_codecs_api",
+      "../api/task_queue:pending_task_safety_flag",
+      "../api/units:time_delta",
+      "../api/video:video_frame",
+      "../api/video:video_rtp_headers",
+      "../api/video_codecs:video_codecs_api",
+      "../media:media_channel",
+      "../media:rtc_media_base",
+      "../p2p:rtc_p2p",
+      "../pc:video_track_source",
+      "../rtc_base:checks",
+      "../rtc_base:logging",
+      "../rtc_base:macromagic",
+      "../rtc_base:net_helpers",
+      "../rtc_base:refcount",
+      "../rtc_base:rtc_certificate_generator",
+      "../rtc_base:ssl",
+      "../rtc_base:stringutils",
+      "../rtc_base:threading",
+      "../rtc_base/third_party/sigslot",
+      "../system_wrappers:field_trial",
+      "../test:field_trial",
+      "../test:platform_video_capturer",
+      "../test:rtp_test_utils",
+      "//third_party/abseil-cpp/absl/memory",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+    if (is_win) {
+      sources += [
+        "peerconnection/client/flag_defs.h",
+        "peerconnection/client/main.cc",
+        "peerconnection/client/main_wnd.cc",
+        "peerconnection/client/main_wnd.h",
+      ]
+      configs += [ "//build/config/win:windowed" ]
+      deps += [
+        "../media:rtc_media_base",
+        "../rtc_base:win32",
+        "../rtc_base:win32_socket_init",
+      ]
+    }
+    if (is_linux || is_chromeos) {
+      sources += [
+        "peerconnection/client/linux/main.cc",
+        "peerconnection/client/linux/main_wnd.cc",
+        "peerconnection/client/linux/main_wnd.h",
+      ]
+      cflags = [ "-Wno-deprecated-declarations" ]
+      libs = [
+        "X11",
+        "Xcomposite",
+        "Xext",
+        "Xrender",
+      ]
+      configs += [ ":gtk_config" ]
+    }
+
+    deps += [
+      "../api:libjingle_peerconnection_api",
+      "../api/audio_codecs:builtin_audio_decoder_factory",
+      "../api/audio_codecs:builtin_audio_encoder_factory",
+      "../api/video:video_frame",
+      "../api/video:video_rtp_headers",
+      "../api/video_codecs:builtin_video_decoder_factory",
+      "../api/video_codecs:builtin_video_encoder_factory",
+      "../media:rtc_audio_video",
+      "../modules/audio_device",
+      "../modules/audio_processing",
+      "../modules/audio_processing:api",
+      "../modules/video_capture:video_capture_module",
+      "../pc:libjingle_peerconnection",
+      "../rtc_base:rtc_json",
+      "../test:video_test_common",
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/flags:parse",
+      "//third_party/libyuv",
+    ]
+  }
+
+  rtc_executable("peerconnection_server") {
+    testonly = true
+    sources = [
+      "peerconnection/server/data_socket.cc",
+      "peerconnection/server/data_socket.h",
+      "peerconnection/server/main.cc",
+      "peerconnection/server/peer_channel.cc",
+      "peerconnection/server/peer_channel.h",
+      "peerconnection/server/utils.cc",
+      "peerconnection/server/utils.h",
+    ]
+    deps = [
+      "../rtc_base:checks",
+      "../rtc_base:stringutils",
+      "../system_wrappers:field_trial",
+      "../test:field_trial",
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/flags:parse",
+      "//third_party/abseil-cpp/absl/flags:usage",
+    ]
+  }
+  rtc_executable("turnserver") {
+    testonly = true
+    sources = [ "turnserver/turnserver_main.cc" ]
+    deps = [
+      ":read_auth_file",
+      "../p2p:p2p_server_utils",
+      "../p2p:rtc_p2p",
+      "../pc:rtc_pc",
+      "../rtc_base:async_udp_socket",
+      "../rtc_base:ip_address",
+      "../rtc_base:socket_address",
+      "../rtc_base:socket_server",
+      "../rtc_base:threading",
+      "//third_party/abseil-cpp/absl/strings:strings",
+    ]
+  }
+  rtc_executable("stunserver") {
+    testonly = true
+    sources = [ "stunserver/stunserver_main.cc" ]
+    deps = [
+      "../p2p:p2p_server_utils",
+      "../p2p:rtc_p2p",
+      "../pc:rtc_pc",
+      "../rtc_base:async_udp_socket",
+      "../rtc_base:socket_address",
+      "../rtc_base:socket_server",
+      "../rtc_base:threading",
+    ]
+  }
+}
+
+if (is_win || is_android) {
+  rtc_shared_library("webrtc_unity_plugin") {
+    testonly = true
+    sources = [
+      "unityplugin/simple_peer_connection.cc",
+      "unityplugin/simple_peer_connection.h",
+      "unityplugin/unity_plugin_apis.cc",
+      "unityplugin/unity_plugin_apis.h",
+      "unityplugin/video_observer.cc",
+      "unityplugin/video_observer.h",
+    ]
+
+    if (is_android) {
+      sources += [
+        "unityplugin/class_reference_holder.cc",
+        "unityplugin/class_reference_holder.h",
+        "unityplugin/jni_onload.cc",
+      ]
+      suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
+    }
+
+    if (is_win) {
+      configs += [ "//build/config/win:windowed" ]
+    }
+    deps = [
+      "../api:create_peerconnection_factory",
+      "../api:libjingle_peerconnection_api",
+      "../api:media_stream_interface",
+      "../api/audio_codecs:builtin_audio_decoder_factory",
+      "../api/audio_codecs:builtin_audio_encoder_factory",
+      "../api/video:video_frame",
+      "../api/video:video_rtp_headers",
+      "../media:rtc_audio_video",
+      "../media:rtc_internal_video_codecs",
+      "../media:rtc_media",
+      "../media:rtc_media_base",
+      "../modules/audio_device",
+      "../modules/audio_processing",
+      "../modules/audio_processing:api",
+      "../modules/video_capture:video_capture_module",
+      "../pc:libjingle_peerconnection",
+      "../pc:video_track_source",
+      "../rtc_base:ssl",
+      "../test:platform_video_capturer",
+      "../test:video_test_common",
+      "//third_party/abseil-cpp/absl/memory",
+    ]
+    if (is_android) {
+      deps += [
+        "../modules/utility",
+        "../sdk/android:libjingle_peerconnection_jni",
+        "../sdk/android:native_api_jni",
+      ]
+    }
+  }
+}
+
+if (is_android) {
+  rtc_android_library("webrtc_unity_java") {
+    sources = [ "unityplugin/java/src/org/webrtc/UnityUtility.java" ]
+    deps = [
+      "../rtc_base:base_java",
+      "../sdk/android:camera_java",
+      "../sdk/android:libjingle_peerconnection_java",
+      "../sdk/android:peerconnection_java",
+      "../sdk/android:video_api_java",
+      "../sdk/android:video_java",
+      "//third_party/androidx:androidx_annotation_annotation_java",
+    ]
+  }
+
+  dist_jar("libwebrtc_unity") {
+    _target_dir_name = get_label_info(":$target_name", "dir")
+    output = "${root_out_dir}/lib.java${_target_dir_name}/${target_name}.jar"
+    direct_deps_only = false
+    use_interface_jars = false
+    use_unprocessed_jars = false
+    requires_android = true
+    deps = [
+      ":webrtc_unity_java",
+      "../rtc_base:base_java",
+      "../sdk/android:libjingle_peerconnection_java",
+      "../sdk/android:libjingle_peerconnection_metrics_default_java",
+      "//third_party/androidx:androidx_annotation_annotation_java",
+    ]
+  }
+
+  robolectric_binary("android_examples_junit_tests") {
+    sources = [
+      "androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java",
+      "androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java",
+      "androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java",
+    ]
+
+    deps = [
+      ":AppRTCMobile_javalib",
+      "../sdk/android:peerconnection_java",
+      "//third_party/androidx:androidx_test_core_java",
+      "//third_party/google-truth:google_truth_java",
+    ]
+
+    additional_jar_files = [ [
+          "../sdk/android/tests/resources/robolectric.properties",
+          "robolectric.properties",
+        ] ]
+  }
+}
+
+if (!build_with_chromium) {
+  # Doesn't build within Chrome on Win.
+  rtc_executable("stun_prober") {
+    testonly = true
+    sources = [ "stunprober/main.cc" ]
+    deps = [
+      "../p2p:libstunprober",
+      "../p2p:rtc_p2p",
+      "../rtc_base:checks",
+      "../rtc_base:logging",
+      "../rtc_base:network",
+      "../rtc_base:socket_address",
+      "../rtc_base:ssl",
+      "../rtc_base:threading",
+      "../rtc_base:timeutils",
+      "../test:scoped_key_value_config",
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/flags:parse",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/examples/DEPS b/third_party/libwebrtc/examples/DEPS
new file mode 100644
index 0000000000..114cda384b
--- /dev/null
+++ b/third_party/libwebrtc/examples/DEPS
@@ -0,0 +1,13 @@
+include_rules = [
+  "+common_video",
+  "+logging/rtc_event_log/rtc_event_log_factory.h",
+  "+media",
+  "+modules/audio_device",
+  "+modules/video_capture",
+  "+modules/audio_processing",
+  "+p2p",
+  "+pc",
+  "+sdk/objc",
+  "+system_wrappers/include",
+  "+third_party/libyuv",
+]
diff --git a/third_party/libwebrtc/examples/OWNERS b/third_party/libwebrtc/examples/OWNERS
new file mode 100644
index 0000000000..ff1f425462
--- /dev/null
+++ b/third_party/libwebrtc/examples/OWNERS
@@ -0,0 +1,4 @@
+magjed@webrtc.org
+perkj@webrtc.org
+tkchin@webrtc.org
+kthelgason@webrtc.org
diff --git a/third_party/libwebrtc/examples/aarproject/.gitignore b/third_party/libwebrtc/examples/aarproject/.gitignore
new file mode 100644
index 0000000000..e93eb885a9
--- /dev/null
+++ b/third_party/libwebrtc/examples/aarproject/.gitignore
@@ -0,0 +1,16 @@
+# Default ignores by Android Studio
+*.iml
+.gradle
+# We want to specify our own SDK.
+# /local.properties
+/.idea/workspace.xml
+/.idea/libraries
+.DS_Store
+/build
+/captures
+.externalNativeBuild
+
+# Additional ignores
+/gradlew
+/gradlew.bat
+/gradle
diff --git a/third_party/libwebrtc/examples/aarproject/OWNERS b/third_party/libwebrtc/examples/aarproject/OWNERS
new file mode 100644
index 0000000000..cf092a316a
--- /dev/null
+++ b/third_party/libwebrtc/examples/aarproject/OWNERS
@@ -0,0 +1 @@
+xalep@webrtc.org
diff --git a/third_party/libwebrtc/examples/aarproject/app/.gitignore b/third_party/libwebrtc/examples/aarproject/app/.gitignore
new file mode 100644
index 0000000000..796b96d1c4
--- /dev/null
+++ b/third_party/libwebrtc/examples/aarproject/app/.gitignore
@@ -0,0 +1 @@
+/build
diff --git a/third_party/libwebrtc/examples/aarproject/app/build.gradle b/third_party/libwebrtc/examples/aarproject/app/build.gradle
new file mode 100644
index 0000000000..94fa851af2
--- /dev/null
+++ b/third_party/libwebrtc/examples/aarproject/app/build.gradle
@@ -0,0 +1,53 @@
+apply plugin: 'com.android.application'
+
+android {
+    compileSdkVersion 31
+    defaultConfig {
+        applicationId "org.appspot.apprtc"
+        minSdkVersion 21
+        targetSdkVersion 31
+        versionCode 1
+        versionName "1.0"
+        testInstrumentationRunner "android.support.test.runner.AndroidJUnitRunner"
+    }
+    buildTypes {
+        release {
+            minifyEnabled false
+            proguardFiles getDefaultProguardFile('proguard-android.txt'), 'proguard-rules.pro'
+        }
+    }
+    compileOptions {
+        sourceCompatibility JavaVersion.VERSION_1_8
+        targetCompatibility JavaVersion.VERSION_1_8
+    }
+    sourceSets {
+        main {
+            manifest.srcFile "../../androidapp/AndroidManifest.xml"
+            java.srcDirs = [
+                    "../../androidapp/src"
+            ]
+            res.srcDirs = [
+                    "../../androidapp/res"
+            ]
+        }
+        androidTest {
+            manifest.srcFile "../../androidtests/AndroidManifest.xml"
+            java.srcDirs = [
+                    "../../androidtests/src"
+            ]
+            // This test doesn't work in Android Studio.
+            java.exclude('**/CallActivityStubbedInputOutputTest.java')
+        }
+    }
+}
+
+dependencies {
+    if (project.hasProperty('aarDir')) {
+        implementation fileTree(dir: project.aarDir, include: ['google-webrtc-*.aar'])
+    }
+    implementation fileTree(dir: '../../androidapp/third_party/autobanh/lib', include: ['autobanh.jar'])
+    implementation 'androidx.annotation:annotation:1.2.0'
+    testImplementation 'junit:junit:4.12'
+    androidTestImplementation 'com.android.support.test:runner:1.0.1'
+    androidTestImplementation 'com.android.support.test.espresso:espresso-core:3.0.1'
+}
diff --git a/third_party/libwebrtc/examples/aarproject/app/proguard-rules.pro b/third_party/libwebrtc/examples/aarproject/app/proguard-rules.pro
new file mode 100644
index 0000000000..d6cc4c5eba
--- /dev/null
+++ b/third_party/libwebrtc/examples/aarproject/app/proguard-rules.pro
@@ -0,0 +1,25 @@
+# Add project specific ProGuard rules here.
+# By default, the flags in this file are appended to flags specified
+# in /usr/local/google/home/sakal/Android/Sdk/tools/proguard/proguard-android.txt
+# You can edit the include path and order by changing the proguardFiles
+# directive in build.gradle.
+#
+# For more details, see
+#   http://developer.android.com/guide/developing/tools/proguard.html
+
+# Add any project specific keep options here:
+
+# If your project uses WebView with JS, uncomment the following
+# and specify the fully qualified class name to the JavaScript interface
+# class:
+#-keepclassmembers class fqcn.of.javascript.interface.for.webview {
+#   public *;
+#}
+
+# Uncomment this to preserve the line number information for
+# debugging stack traces.
+#-keepattributes SourceFile,LineNumberTable
+
+# If you keep the line number information, uncomment this to
+# hide the original source file name.
+#-renamesourcefileattribute SourceFile
diff --git a/third_party/libwebrtc/examples/aarproject/build.gradle b/third_party/libwebrtc/examples/aarproject/build.gradle
new file mode 100644
index 0000000000..969effd3ed
--- /dev/null
+++ b/third_party/libwebrtc/examples/aarproject/build.gradle
@@ -0,0 +1,27 @@
+// Top-level build file where you can add configuration options common to all sub-projects/modules.
+
+buildscript {
+
+    repositories {
+        google()
+        mavenCentral()
+    }
+    dependencies {
+        classpath "com.android.tools.build:gradle:7.0.3"
+
+
+        // NOTE: Do not place your application dependencies here; they belong
+        // in the individual module build.gradle files
+    }
+}
+
+allprojects {
+    repositories {
+        google()
+        mavenCentral()
+    }
+}
+
+task clean(type: Delete) {
+    delete rootProject.buildDir
+}
diff --git a/third_party/libwebrtc/examples/aarproject/gradle.properties b/third_party/libwebrtc/examples/aarproject/gradle.properties
new file mode 100644
index 0000000000..2e87c52f83
--- /dev/null
+++ b/third_party/libwebrtc/examples/aarproject/gradle.properties
@@ -0,0 +1,22 @@
+# Project-wide Gradle settings.
+
+# IDE (e.g. Android Studio) users:
+# Gradle settings configured through the IDE *will override*
+# any settings specified in this file.
+
+# For more details on how to configure your build environment visit
+# http://www.gradle.org/docs/current/userguide/build_environment.html
+
+# Specifies the JVM arguments used for the daemon process.
+# The setting is particularly useful for tweaking memory settings.
+org.gradle.jvmargs=-Xmx1536m
+
+# When configured, Gradle will run in incubating parallel mode.
+# This option should only be used with decoupled projects. More details, visit
+# http://www.gradle.org/docs/current/userguide/multi_project_builds.html#sec:decoupled_projects
+# org.gradle.parallel=true
+
+# AndroidX package structure to make it clearer which packages are bundled with the
+# Android operating system, and which are packaged with your app's APK
+# https://developer.android.com/topic/libraries/support-library/androidx-rn
+android.useAndroidX=true
diff --git a/third_party/libwebrtc/examples/aarproject/local.properties b/third_party/libwebrtc/examples/aarproject/local.properties
new file mode 100644
index 0000000000..99cdcd2674
--- /dev/null
+++ b/third_party/libwebrtc/examples/aarproject/local.properties
@@ -0,0 +1,2 @@
+# Use Android SDK from third_party/android_sdk/public
+sdk.dir=../../third_party/android_sdk/public
diff --git a/third_party/libwebrtc/examples/aarproject/settings.gradle b/third_party/libwebrtc/examples/aarproject/settings.gradle
new file mode 100644
index 0000000000..e7b4def49c
--- /dev/null
+++ b/third_party/libwebrtc/examples/aarproject/settings.gradle
@@ -0,0 +1 @@
+include ':app'
diff --git a/third_party/libwebrtc/examples/androidapp/AndroidManifest.xml b/third_party/libwebrtc/examples/androidapp/AndroidManifest.xml
new file mode 100644
index 0000000000..05f1bd3da3
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/AndroidManifest.xml
@@ -0,0 +1,60 @@
+<?xml version="1.0" encoding="utf-8"?>
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+          xmlns:tools="http://schemas.android.com/tools"
+          package="org.appspot.apprtc"
+          android:versionCode="1"
+          android:versionName="1.0">
+
+    <uses-feature android:name="android.hardware.camera" />
+    <uses-feature android:name="android.hardware.camera.autofocus" />
+    <uses-feature android:glEsVersion="0x00020000" android:required="true" />
+
+    <uses-permission android:name="android.permission.CAMERA" />
+    <uses-permission android:name="android.permission.CHANGE_NETWORK_STATE" />
+    <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
+    <uses-permission android:name="android.permission.RECORD_AUDIO" />
+    <uses-permission android:name="android.permission.BLUETOOTH" />
+    <uses-permission android:name="android.permission.INTERNET" />
+    <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
+    <uses-permission android:name="android.permission.ACCESS_NETWORK_STATE" />
+
+    <!-- This is a test application that should always be debuggable. -->
+    <application android:label="@string/app_name"
+                 android:icon="@drawable/ic_launcher"
+                 android:allowBackup="false"
+                 android:debuggable="true"
+                 android:supportsRtl="false"
+                 tools:ignore="HardcodedDebugMode">
+
+        <activity android:name="ConnectActivity"
+                  android:label="@string/app_name"
+                  android:windowSoftInputMode="adjustPan"
+                  android:theme="@style/AppTheme"
+                  android:exported="true">
+            <intent-filter>
+                <action android:name="android.intent.action.MAIN"/>
+                <category android:name="android.intent.category.LAUNCHER"/>
+            </intent-filter>
+
+            <intent-filter>
+                <action android:name="android.intent.action.VIEW"/>
+                <category android:name="android.intent.category.DEFAULT"/>
+                <category android:name="android.intent.category.BROWSABLE"/>
+                <data android:scheme="https" android:host="appr.tc"/>
+                <data android:scheme="http" android:host="appr.tc"/>
+            </intent-filter>
+        </activity>
+
+        <activity android:name="SettingsActivity"
+                  android:label="@string/settings_name"
+                  android:theme="@style/AppTheme">
+        </activity>
+
+        <activity android:name="CallActivity"
+                  android:label="@string/app_name"
+                  android:screenOrientation="fullUser"
+                  android:configChanges="orientation|smallestScreenSize|screenSize|screenLayout"
+                  android:theme="@style/CallActivityTheme">
+        </activity>
+    </application>
+</manifest>
diff --git a/third_party/libwebrtc/examples/androidapp/OWNERS b/third_party/libwebrtc/examples/androidapp/OWNERS
new file mode 100644
index 0000000000..109bea2725
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/OWNERS
@@ -0,0 +1,2 @@
+magjed@webrtc.org
+xalep@webrtc.org
diff --git a/third_party/libwebrtc/examples/androidapp/README b/third_party/libwebrtc/examples/androidapp/README
new file mode 100644
index 0000000000..97e609117c
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/README
@@ -0,0 +1,23 @@
+This directory contains an example Android client for https://appr.tc
+
+Prerequisites:
+- "Getting the code", "Compiling", and "Using the Bundled Android SDK/NDK"
+  on http://www.webrtc.org/native-code/android
+
+Example of building & using the app:
+
+cd <path/to/webrtc>/src
+ninja -C out/Default AppRTCMobile
+adb install -r out/Default/apks/AppRTCMobile.apk
+
+In desktop chrome, navigate to https://appr.tc and note the r=<NNN> room
+this redirects to or navigate directly to https://appr.tc/r/<NNN> with
+your own room number. Launch AppRTC on the device and add same <NNN> into the room name list.
+
+You can also run application from a command line to connect to the first room in a list:
+adb shell am start -n org.appspot.apprtc/.ConnectActivity -a android.intent.action.VIEW
+This should result in the app launching on Android and connecting to the 3-dot-apprtc
+page displayed in the desktop browser.
+To run loopback test execute following command:
+adb shell am start -n org.appspot.apprtc/.ConnectActivity -a android.intent.action.VIEW --ez "org.appspot.apprtc.LOOPBACK" true
+
diff --git a/third_party/libwebrtc/examples/androidapp/ant.properties b/third_party/libwebrtc/examples/androidapp/ant.properties
new file mode 100644
index 0000000000..b0971e891e
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/ant.properties
@@ -0,0 +1,17 @@
+# This file is used to override default values used by the Ant build system.
+#
+# This file must be checked into Version Control Systems, as it is
+# integral to the build system of your project.
+
+# This file is only used by the Ant script.
+
+# You can use this to override default values such as
+#  'source.dir' for the location of your java source folder and
+#  'out.dir' for the location of your output folder.
+
+# You can also use it define how the release builds are signed by declaring
+# the following properties:
+#  'key.store' for the location of your keystore and
+#  'key.alias' for the name of the key to use.
+# The password will be asked during the build when you use the 'release' target.
+
diff --git a/third_party/libwebrtc/examples/androidapp/build.xml b/third_party/libwebrtc/examples/androidapp/build.xml
new file mode 100644
index 0000000000..aa1db6db79
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/build.xml
@@ -0,0 +1,92 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<project name="AppRTCMobile" default="help">
+
+    <!-- The local.properties file is created and updated by the 'android' tool.
+         It contains the path to the SDK. It should *NOT* be checked into
+         Version Control Systems. -->
+    <property file="local.properties" />
+
+    <!-- The ant.properties file can be created by you. It is only edited by the
+         'android' tool to add properties to it.
+         This is the place to change some Ant specific build properties.
+         Here are some properties you may want to change/update:
+
+         source.dir
+             The name of the source directory. Default is 'src'.
+         out.dir
+             The name of the output directory. Default is 'bin'.
+
+         For other overridable properties, look at the beginning of the rules
+         files in the SDK, at tools/ant/build.xml
+
+         Properties related to the SDK location or the project target should
+         be updated using the 'android' tool with the 'update' action.
+
+         This file is an integral part of the build system for your
+         application and should be checked into Version Control Systems.
+
+         -->
+    <property file="ant.properties" />
+
+    <!-- if sdk.dir was not set from one of the property file, then
+         get it from the ANDROID_HOME env var.
+         This must be done before we load project.properties since
+         the proguard config can use sdk.dir -->
+    <property environment="env" />
+    <condition property="sdk.dir" value="${env.ANDROID_SDK_ROOT}">
+        <isset property="env.ANDROID_SDK_ROOT" />
+    </condition>
+
+    <!-- The project.properties file is created and updated by the 'android'
+         tool, as well as ADT.
+
+         This contains project specific properties such as project target, and library
+         dependencies. Lower level build properties are stored in ant.properties
+         (or in .classpath for Eclipse projects).
+
+         This file is an integral part of the build system for your
+         application and should be checked into Version Control Systems. -->
+    <loadproperties srcFile="project.properties" />
+
+    <!-- quick check on sdk.dir -->
+    <fail
+            message="sdk.dir is missing. Make sure to generate local.properties using 'android update project' or to inject it through the ANDROID_HOME environment variable."
+            unless="sdk.dir"
+    />
+
+    <!--
+        Import per project custom build rules if present at the root of the project.
+        This is the place to put custom intermediary targets such as:
+            -pre-build
+            -pre-compile
+            -post-compile (This is typically used for code obfuscation.
+                           Compiled code location: ${out.classes.absolute.dir}
+                           If this is not done in place, override ${out.dex.input.absolute.dir})
+            -post-package
+            -post-build
+            -pre-clean
+    -->
+    <import file="custom_rules.xml" optional="true" />
+
+    <!-- Import the actual build file.
+
+         To customize existing targets, there are two options:
+         - Customize only one target:
+             - copy/paste the target into this file, *before* the
+               <import> task.
+             - customize it to your needs.
+         - Customize the whole content of build.xml
+             - copy/paste the content of the rules files (minus the top node)
+               into this file, replacing the <import> task.
+             - customize to your needs.
+
+         ***********************
+         ****** IMPORTANT ******
+         ***********************
+         In all cases you must update the value of version-tag below to read 'custom' instead of an integer,
+         in order to avoid having your file be overridden by tools such as "android update project"
+    -->
+    <!-- version-tag: 1 -->
+    <import file="${sdk.dir}/tools/ant/build.xml" />
+
+</project>
diff --git a/third_party/libwebrtc/examples/androidapp/project.properties b/third_party/libwebrtc/examples/androidapp/project.properties
new file mode 100644
index 0000000000..a6ca533fe3
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/project.properties
@@ -0,0 +1,16 @@
+# This file is automatically generated by Android Tools.
+# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
+#
+# This file must be checked in Version Control Systems.
+#
+# To customize properties used by the Ant build system edit
+# "ant.properties", and override values to adapt the script to your
+# project structure.
+#
+# To enable ProGuard to shrink and obfuscate your code, uncomment this (available properties: sdk.dir, user.home):
+#proguard.config=${sdk.dir}/tools/proguard/proguard-android.txt:proguard-project.txt
+
+# Project target.
+target=android-22
+
+java.compilerargs=-Xlint:all -Werror
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/disconnect.png b/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/disconnect.png
new file mode 100644
index 0000000000..be36174c24
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/disconnect.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_action_full_screen.png b/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_action_full_screen.png
new file mode 100644
index 0000000000..22f30d31ca
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_action_full_screen.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_action_return_from_full_screen.png b/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_action_return_from_full_screen.png
new file mode 100644
index 0000000000..d9436e5248
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_action_return_from_full_screen.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_launcher.png b/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_launcher.png
new file mode 100644
index 0000000000..f01a31a717
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_launcher.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_loopback_call.png b/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_loopback_call.png
new file mode 100644
index 0000000000..39311853b3
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-hdpi/ic_loopback_call.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/disconnect.png b/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/disconnect.png
new file mode 100644
index 0000000000..be36174c24
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/disconnect.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/ic_action_full_screen.png b/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/ic_action_full_screen.png
new file mode 100644
index 0000000000..e4a9ff0a8e
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/ic_action_full_screen.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/ic_action_return_from_full_screen.png b/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/ic_action_return_from_full_screen.png
new file mode 100644
index 0000000000..f5c80f00e7
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/ic_action_return_from_full_screen.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/ic_launcher.png b/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/ic_launcher.png
new file mode 100644
index 0000000000..5492ed770a
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/ic_launcher.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/ic_loopback_call.png b/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/ic_loopback_call.png
new file mode 100644
index 0000000000..39311853b3
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-ldpi/ic_loopback_call.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/disconnect.png b/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/disconnect.png
new file mode 100644
index 0000000000..be36174c24
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/disconnect.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/ic_action_full_screen.png b/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/ic_action_full_screen.png
new file mode 100644
index 0000000000..e4a9ff0a8e
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/ic_action_full_screen.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/ic_action_return_from_full_screen.png b/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/ic_action_return_from_full_screen.png
new file mode 100644
index 0000000000..f5c80f00e7
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/ic_action_return_from_full_screen.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/ic_launcher.png b/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/ic_launcher.png
new file mode 100644
index 0000000000..b8b4b0ec4b
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/ic_launcher.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/ic_loopback_call.png b/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/ic_loopback_call.png
new file mode 100644
index 0000000000..39311853b3
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-mdpi/ic_loopback_call.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/disconnect.png b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/disconnect.png
new file mode 100644
index 0000000000..be36174c24
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/disconnect.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_action_full_screen.png b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_action_full_screen.png
new file mode 100644
index 0000000000..6d90c071d5
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_action_full_screen.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_action_return_from_full_screen.png b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_action_return_from_full_screen.png
new file mode 100644
index 0000000000..a773b34208
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_action_return_from_full_screen.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_launcher.png b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_launcher.png
new file mode 100644
index 0000000000..a3cd45890c
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_launcher.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_loopback_call.png b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_loopback_call.png
new file mode 100644
index 0000000000..39311853b3
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/drawable-xhdpi/ic_loopback_call.png
diff --git a/third_party/libwebrtc/examples/androidapp/res/layout/activity_call.xml b/third_party/libwebrtc/examples/androidapp/res/layout/activity_call.xml
new file mode 100644
index 0000000000..bf811426f3
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/layout/activity_call.xml
@@ -0,0 +1,34 @@
+<?xml version="1.0" encoding="utf-8"?>
+
+<!-- tools:ignore is needed because lint thinks this can be replaced with a merge. Replacing this
+     with a merge causes the fullscreen SurfaceView not to be centered. -->
+<FrameLayout
+    xmlns:android="http://schemas.android.com/apk/res/android"
+    xmlns:tools="http://schemas.android.com/tools"
+    android:layout_width="match_parent"
+    android:layout_height="match_parent"
+    tools:ignore="MergeRootFrame">
+
+    <org.webrtc.SurfaceViewRenderer
+        android:id="@+id/fullscreen_video_view"
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_gravity="center" />
+
+    <org.webrtc.SurfaceViewRenderer
+        android:id="@+id/pip_video_view"
+        android:layout_height="144dp"
+        android:layout_width="wrap_content"
+        android:layout_gravity="bottom|end"
+        android:layout_margin="16dp"/>
+
+    <FrameLayout
+        android:id="@+id/call_fragment_container"
+        android:layout_width="match_parent"
+        android:layout_height="match_parent" />
+    <FrameLayout
+        android:id="@+id/hud_fragment_container"
+        android:layout_width="match_parent"
+        android:layout_height="match_parent" />
+
+</FrameLayout>
diff --git a/third_party/libwebrtc/examples/androidapp/res/layout/activity_connect.xml b/third_party/libwebrtc/examples/androidapp/res/layout/activity_connect.xml
new file mode 100644
index 0000000000..017e5cabff
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/layout/activity_connect.xml
@@ -0,0 +1,80 @@
+<?xml version="1.0" encoding="utf-8"?>
+<LinearLayout
+    xmlns:android="http://schemas.android.com/apk/res/android"
+    xmlns:tools="http://schemas.android.com/tools"
+    android:layout_margin="16dp"
+    android:layout_width="match_parent"
+    android:layout_height="match_parent"
+    android:orientation="vertical"
+    android:weightSum="1"
+    android:layout_centerHorizontal="true">
+
+    <TextView
+        android:id="@+id/room_edittext_description"
+        android:layout_width="match_parent"
+        android:layout_height="wrap_content"
+        android:text="@string/room_description"/>
+
+    <LinearLayout
+        android:orientation="horizontal"
+        android:layout_width="match_parent"
+        android:layout_height="wrap_content"
+        android:gravity="center"
+        android:layout_marginBottom="8dp">
+
+        <!-- TODO(crbug.com/900912): Fix and remove lint ignore -->
+        <EditText
+            tools:ignore="LabelFor,Autofill"
+            android:id="@+id/room_edittext"
+            android:layout_width="0dp"
+            android:layout_height="wrap_content"
+            android:layout_weight="1"
+            android:maxLines="1"
+            android:imeOptions="actionDone"
+            android:inputType="text"/>
+
+        <ImageButton
+            android:id="@+id/connect_button"
+            android:layout_width="48dp"
+            android:layout_height="48dp"
+            android:contentDescription="@string/connect_description"
+            android:background="@android:drawable/sym_action_call" />
+
+        <ImageButton
+            android:id="@+id/add_favorite_button"
+            android:layout_width="48dp"
+            android:layout_height="48dp"
+            android:contentDescription="@string/add_favorite_description"
+            android:background="@android:drawable/ic_input_add" />
+    </LinearLayout>
+
+    <TextView
+        android:id="@+id/room_listview_description"
+        android:layout_width="match_parent"
+        android:layout_height="48dp"
+        android:layout_marginTop="8dp"
+        android:lines="1"
+        android:maxLines="1"
+        android:textAppearance="?android:attr/textAppearanceMedium"
+        android:text="@string/favorites"
+        android:gravity="center_vertical"/>
+
+    <FrameLayout
+        android:layout_width="match_parent"
+        android:layout_height="0dp"
+        android:layout_weight="1">
+
+        <ListView
+            android:id="@+id/room_listview"
+            android:layout_width="match_parent"
+            android:layout_height="match_parent"
+            android:drawSelectorOnTop="false" />
+
+        <TextView
+            android:id="@android:id/empty"
+            android:layout_width="match_parent"
+            android:layout_height="match_parent"
+            android:gravity="center"
+            android:text="@string/no_favorites" />
+    </FrameLayout>
+</LinearLayout>
diff --git a/third_party/libwebrtc/examples/androidapp/res/layout/fragment_call.xml b/third_party/libwebrtc/examples/androidapp/res/layout/fragment_call.xml
new file mode 100644
index 0000000000..90b1e9ca0e
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/layout/fragment_call.xml
@@ -0,0 +1,77 @@
+<?xml version="1.0" encoding="utf-8"?>
+
+<RelativeLayout
+    xmlns:android="http://schemas.android.com/apk/res/android"
+    android:layout_width="match_parent"
+    android:layout_height="match_parent">
+
+    <TextView
+        android:id="@+id/contact_name_call"
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_centerHorizontal="true"
+        android:layout_above="@+id/buttons_call_container"
+        android:textSize="24sp"
+        android:layout_margin="8dp"/>
+
+    <LinearLayout
+           android:id="@+id/buttons_call_container"
+           android:orientation="horizontal"
+           android:layout_above="@+id/capture_format_text_call"
+           android:layout_alignWithParentIfMissing="true"
+           android:layout_marginBottom="32dp"
+           android:layout_centerHorizontal="true"
+           android:layout_width="wrap_content"
+           android:layout_height="wrap_content">
+
+       <ImageButton
+           android:id="@+id/button_call_disconnect"
+           android:background="@drawable/disconnect"
+           android:contentDescription="@string/disconnect_call"
+           android:layout_marginEnd="16dp"
+           android:layout_width="48dp"
+           android:layout_height="48dp"/>
+
+       <ImageButton
+           android:id="@+id/button_call_switch_camera"
+           android:background="@android:drawable/ic_menu_camera"
+           android:contentDescription="@string/switch_camera"
+           android:layout_marginEnd="8dp"
+           android:layout_width="48dp"
+           android:layout_height="48dp"/>
+
+        <ImageButton
+           android:id="@+id/button_call_scaling_mode"
+           android:background="@drawable/ic_action_return_from_full_screen"
+           android:contentDescription="@string/disconnect_call"
+           android:layout_width="48dp"
+           android:layout_height="48dp"/>
+
+       <ImageButton
+           android:id="@+id/button_call_toggle_mic"
+           android:background="@android:drawable/ic_btn_speak_now"
+           android:contentDescription="@string/toggle_mic"
+           android:layout_marginEnd="8dp"
+           android:layout_width="48dp"
+           android:layout_height="48dp"/>
+    </LinearLayout>
+
+    <TextView
+        android:id="@+id/capture_format_text_call"
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_centerHorizontal="true"
+        android:layout_above="@+id/capture_format_slider_call"
+        android:textSize="16sp"
+        android:text="@string/capture_format_change_text"/>
+
+    <SeekBar
+        android:id="@+id/capture_format_slider_call"
+        android:layout_width="match_parent"
+        android:layout_height="wrap_content"
+        android:layout_centerHorizontal="true"
+        android:layout_alignParentBottom="true"
+        android:progress="50"
+        android:layout_margin="8dp"/>
+
+</RelativeLayout>
diff --git a/third_party/libwebrtc/examples/androidapp/res/layout/fragment_hud.xml b/third_party/libwebrtc/examples/androidapp/res/layout/fragment_hud.xml
new file mode 100644
index 0000000000..483e7ba456
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/layout/fragment_hud.xml
@@ -0,0 +1,27 @@
+<?xml version="1.0" encoding="utf-8"?>
+
+<RelativeLayout
+    xmlns:android="http://schemas.android.com/apk/res/android"
+    android:layout_width="match_parent"
+    android:layout_height="match_parent">
+
+    <ImageButton
+        android:id="@+id/button_toggle_debug"
+        android:background="@android:drawable/ic_menu_info_details"
+        android:contentDescription="@string/toggle_debug"
+        android:layout_alignParentBottom="true"
+        android:layout_alignParentStart="true"
+        android:layout_width="48dp"
+        android:layout_height="48dp"/>
+
+    <TextView
+        android:id="@+id/hud_stat_call"
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_alignParentEnd="true"
+        android:textStyle="bold"
+        android:textColor="#C000FF00"
+        android:textSize="12sp"
+        android:layout_margin="8dp"/>
+
+</RelativeLayout>
diff --git a/third_party/libwebrtc/examples/androidapp/res/menu/connect_menu.xml b/third_party/libwebrtc/examples/androidapp/res/menu/connect_menu.xml
new file mode 100644
index 0000000000..a723f54941
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/menu/connect_menu.xml
@@ -0,0 +1,13 @@
+<menu xmlns:android="http://schemas.android.com/apk/res/android">
+    <item
+        android:id="@+id/action_loopback"
+        android:icon="@drawable/ic_loopback_call"
+        android:showAsAction="always"
+        android:title="@string/action_loopback"/>
+    <item
+        android:id="@+id/action_settings"
+        android:orderInCategory="100"
+        android:icon="@android:drawable/ic_menu_preferences"
+        android:showAsAction="ifRoom"
+        android:title="@string/action_settings"/>
+</menu>
diff --git a/third_party/libwebrtc/examples/androidapp/res/values-v17/styles.xml b/third_party/libwebrtc/examples/androidapp/res/values-v17/styles.xml
new file mode 100644
index 0000000000..969b5012e9
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/values-v17/styles.xml
@@ -0,0 +1,10 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+  <style name="AppTheme" parent="android:Theme.Holo" />
+
+  <style name="CallActivityTheme" parent="android:Theme.Black">
+    <item name="android:windowActionBar">false</item>
+    <item name="android:windowFullscreen">true</item>
+    <item name="android:windowNoTitle">true</item>
+  </style>
+</resources>
diff --git a/third_party/libwebrtc/examples/androidapp/res/values-v21/styles.xml b/third_party/libwebrtc/examples/androidapp/res/values-v21/styles.xml
new file mode 100644
index 0000000000..b19af7e38f
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/values-v21/styles.xml
@@ -0,0 +1,4 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+  <style name="AppTheme" parent="android:Theme.Material" />
+</resources>
diff --git a/third_party/libwebrtc/examples/androidapp/res/values/arrays.xml b/third_party/libwebrtc/examples/androidapp/res/values/arrays.xml
new file mode 100644
index 0000000000..4a2948c875
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/values/arrays.xml
@@ -0,0 +1,61 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+    <string-array name="videoResolutions">
+        <item>Default</item>
+        <item>4K (3840 x 2160)</item>
+        <item>Full HD (1920 x 1080)</item>
+        <item>HD (1280 x 720)</item>
+        <item>VGA (640 x 480)</item>
+        <item>QVGA (320 x 240)</item>
+    </string-array>
+
+    <string-array name="videoResolutionsValues">
+        <item>Default</item>
+        <item>3840 x 2160</item>
+        <item>1920 x 1080</item>
+        <item>1280 x 720</item>
+        <item>640 x 480</item>
+        <item>320 x 240</item>
+    </string-array>
+
+    <string-array name="cameraFps">
+        <item>Default</item>
+        <item>30 fps</item>
+        <item>15 fps</item>
+    </string-array>
+
+    <string-array name="startBitrate">
+        <item>Default</item>
+        <item>Manual</item>
+    </string-array>
+
+    <string-array name="videoCodecs">
+        <item>VP8</item>
+        <item>VP9</item>
+        <item>H264 Baseline</item>
+        <item>H264 High</item>
+        <item>AV1</item>
+    </string-array>
+
+    <string-array name="audioCodecs">
+        <item>OPUS</item>
+        <item>ISAC</item>
+    </string-array>
+
+    <string-array name="speakerphone">
+        <item>Auto (proximity sensor)</item>
+        <item>Enabled</item>
+        <item>Disabled</item>
+    </string-array>
+
+    <string-array name="speakerphoneValues">
+        <item>auto</item>
+        <item>true</item>
+        <item>false</item>
+    </string-array>
+
+    <string-array name="roomListContextMenu">
+        <item>Remove favorite</item>
+    </string-array>
+
+</resources>
diff --git a/third_party/libwebrtc/examples/androidapp/res/values/strings.xml b/third_party/libwebrtc/examples/androidapp/res/values/strings.xml
new file mode 100644
index 0000000000..814966f200
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/values/strings.xml
@@ -0,0 +1,224 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+    <string name="app_name" translatable="false">AppRTC</string>
+    <string name="settings_name" translatable="false">AppRTC Settings</string>
+    <string name="disconnect_call">Disconnect Call</string>
+    <string name="room_description">
+        Please enter a room name. Room names are shared with everyone, so think
+        of something unique and send it to a friend.
+    </string>
+    <string name="favorites">Favorites</string>
+    <string name="no_favorites">No favorites</string>
+    <string name="invalid_url_title">Invalid URL</string>
+    <string name="invalid_url_text">The URL or room name you entered resulted in an invalid URL: %1$s
+    </string>
+    <string name="channel_error_title">Connection error</string>
+    <string name="connecting_to">Connecting to: %1$s</string>
+    <string name="missing_url">FATAL ERROR: Missing URL to connect to.</string>
+    <string name="camera2_texture_only_error">Camera2 only supports capturing to texture. Either disable Camera2 or enable capturing to texture in the options.</string>
+    <string name="ok">OK</string>
+    <string name="switch_camera">Switch front/back camera</string>
+    <string name="capture_format_change_text">Slide to change capture format</string>
+    <string name="muted">Muted</string>
+    <string name="toggle_debug">Toggle debug view</string>
+    <string name="toggle_mic">Toggle microphone on/off</string>
+    <string name="action_settings">Settings</string>
+    <string name="action_loopback">Loopback connection</string>
+    <string name="connect_description">Connect to the room</string>
+    <string name="add_favorite_description">Add favorite</string>
+    <string name="format_description">%1$dx%2$d @ %3$d fps</string>
+    <string name="missing_permissions_try_again">The application is missing permissions. It might not work correctly. Do you want to try again?</string>
+    <string name="yes">Yes</string>
+    <string name="no">No</string>
+
+    <!-- Settings strings. -->
+    <string name="pref_room_key">room_preference</string>
+    <string name="pref_room_list_key">room_list_preference</string>
+
+    <string name="pref_videosettings_key">video_settings_key</string>
+    <string name="pref_videosettings_title">WebRTC video settings.</string>
+
+    <string name="pref_videocall_key">videocall_preference</string>
+    <string name="pref_videocall_title">Video call.</string>
+    <string name="pref_videocall_dlg">Enable video in a call.</string>
+    <string name="pref_videocall_default">true</string>
+
+    <string name="pref_screencapture_key">screencapture_preference</string>
+    <string name="pref_screencapture_title">Use screencapture.</string>
+    <string name="pref_screencapture_default">false</string>
+
+    <string name="pref_camera2_key">camera2_preference</string>
+    <string name="pref_camera2_title">Use Camera2.</string>
+    <string name="pref_camera2_default">true</string>
+    <string name="pref_camera2_not_supported">Not supported on this device.</string>
+
+    <string name="pref_resolution_key">resolution_preference</string>
+    <string name="pref_resolution_title">Video resolution.</string>
+    <string name="pref_resolution_dlg">Enter AppRTC local video resolution.</string>
+    <string name="pref_resolution_default">Default</string>
+
+    <string name="pref_fps_key">fps_preference</string>
+    <string name="pref_fps_title">Camera fps.</string>
+    <string name="pref_fps_dlg">Enter local camera fps.</string>
+    <string name="pref_fps_default">Default</string>
+
+    <string name="pref_capturequalityslider_key">capturequalityslider_preference</string>
+    <string name="pref_capturequalityslider_title">Capture quality slider.</string>
+    <string name="pref_capturequalityslider_dlg">Enable slider for changing capture quality.</string>
+    <string name="pref_capturequalityslider_default">false</string>
+
+    <string name="pref_maxvideobitrate_key">maxvideobitrate_preference</string>
+    <string name="pref_maxvideobitrate_title">Maximum video bitrate setting.</string>
+    <string name="pref_maxvideobitrate_dlg">Maximum video bitrate setting.</string>
+    <string name="pref_maxvideobitrate_default">Default</string>
+
+    <string name="pref_maxvideobitratevalue_key">maxvideobitratevalue_preference</string>
+    <string name="pref_maxvideobitratevalue_title">Video encoder maximum bitrate.</string>
+    <string name="pref_maxvideobitratevalue_dlg">Enter video encoder maximum bitrate in kbps.</string>
+    <string name="pref_maxvideobitratevalue_default">1700</string>
+
+    <string name="pref_videocodec_key">videocodec_preference</string>
+    <string name="pref_videocodec_title">Default video codec.</string>
+    <string name="pref_videocodec_dlg">Select default video codec.</string>
+    <string name="pref_videocodec_default">VP8</string>
+
+    <string name="pref_hwcodec_key">hwcodec_preference</string>
+    <string name="pref_hwcodec_title">Video codec hardware acceleration.</string>
+    <string name="pref_hwcodec_dlg">Use hardware accelerated video codec (if available).</string>
+    <string name="pref_hwcodec_default">true</string>
+
+    <string name="pref_capturetotexture_key">capturetotexture_preference</string>
+    <string name="pref_capturetotexture_title">Video capture to surface texture.</string>
+    <string name="pref_capturetotexture_dlg">Capture video to textures (if available).</string>
+    <string name="pref_capturetotexture_default">true</string>
+
+    <string name="pref_flexfec_key">flexfec_preference</string>
+    <string name="pref_flexfec_title">Codec-agnostic Flexible FEC.</string>
+    <string name="pref_flexfec_dlg">Enable FlexFEC.</string>
+    <string name="pref_flexfec_default">false</string>
+
+    <string name="pref_value_enabled">Enabled</string>
+    <string name="pref_value_disabled">Disabled</string>
+
+    <string name="pref_audiosettings_key">audio_settings_key</string>
+    <string name="pref_audiosettings_title">WebRTC audio settings.</string>
+
+    <string name="pref_startaudiobitrate_key">startaudiobitrate_preference</string>
+    <string name="pref_startaudiobitrate_title">Audio bitrate setting.</string>
+    <string name="pref_startaudiobitrate_dlg">Audio bitrate setting.</string>
+    <string name="pref_startaudiobitrate_default">Default</string>
+
+    <string name="pref_startaudiobitratevalue_key">startaudiobitratevalue_preference</string>
+    <string name="pref_startaudiobitratevalue_title">Audio codec bitrate.</string>
+    <string name="pref_startaudiobitratevalue_dlg">Enter audio codec bitrate in kbps.</string>
+    <string name="pref_startaudiobitratevalue_default">32</string>
+
+    <string name="pref_audiocodec_key">audiocodec_preference</string>
+    <string name="pref_audiocodec_title">Default audio codec.</string>
+    <string name="pref_audiocodec_dlg">Select default audio codec.</string>
+    <string name="pref_audiocodec_default">OPUS</string>
+
+    <string name="pref_noaudioprocessing_key">audioprocessing_preference</string>
+    <string name="pref_noaudioprocessing_title">Disable audio processing.</string>
+    <string name="pref_noaudioprocessing_dlg">Disable audio processing pipeline.</string>
+    <string name="pref_noaudioprocessing_default">false</string>
+
+    <string name="pref_aecdump_key">aecdump_preference</string>
+    <string name="pref_aecdump_title">Create aecdump.</string>
+    <string name="pref_aecdump_dlg">Enable diagnostic audio recordings.</string>
+    <string name="pref_aecdump_default">false</string>
+
+    <string name="pref_enable_save_input_audio_to_file_key">enable_key</string>
+    <string name="pref_enable_save_input_audio_to_file_title">Save input audio to file.</string>
+    <string name="pref_enable_save_input_audio_to_file_dlg">Save input audio to file.</string>
+    <string name="pref_enable_save_input_audio_to_file_default">false</string>
+
+    <string name="pref_opensles_key">opensles_preference</string>
+    <string name="pref_opensles_title">Use OpenSL ES for audio playback.</string>
+    <string name="pref_opensles_dlg">Use OpenSL ES for audio playback.</string>
+    <string name="pref_opensles_default">false</string>
+
+    <string name="pref_disable_built_in_aec_key">disable_built_in_aec_preference</string>
+    <string name="pref_disable_built_in_aec_title">Disable hardware AEC.</string>
+    <string name="pref_disable_built_in_aec_dlg">Disable hardware AEC.</string>
+    <string name="pref_disable_built_in_aec_default">false</string>
+    <string name="pref_built_in_aec_not_available">Hardware AEC is not available</string>
+
+    <string name="pref_disable_built_in_agc_key">disable_built_in_agc_preference</string>
+    <string name="pref_disable_built_in_agc_title">Disable hardware AGC.</string>
+    <string name="pref_disable_built_in_agc_dlg">Disable hardware AGC.</string>
+    <string name="pref_disable_built_in_agc_default">false</string>
+    <string name="pref_built_in_agc_not_available">Hardware AGC is not available</string>
+
+    <string name="pref_disable_built_in_ns_key">disable_built_in_ns_preference</string>
+    <string name="pref_disable_built_in_ns_title">Disable hardware NS.</string>
+    <string name="pref_disable_built_in_ns_dlg">Disable hardware NS.</string>
+    <string name="pref_disable_built_in_ns_default">false</string>
+    <string name="pref_built_in_ns_not_available">Hardware NS is not available</string>
+
+    <string name="pref_disable_webrtc_agc_and_hpf_key">disable_webrtc_agc_and_hpf_preference</string>
+    <string name="pref_disable_webrtc_agc_and_hpf_title">Disable WebRTC AGC and HPF.</string>
+    <string name="pref_disable_webrtc_agc_default">false</string>
+
+    <string name="pref_speakerphone_key">speakerphone_preference</string>
+    <string name="pref_speakerphone_title">Speakerphone.</string>
+    <string name="pref_speakerphone_dlg">Speakerphone.</string>
+    <string name="pref_speakerphone_default">auto</string>
+
+    <string name="pref_datasettings_key">data_settings_key</string>
+    <string name="pref_datasettings_title">WebRTC data channel settings.</string>
+
+    <string name="pref_enable_datachannel_key">enable_datachannel_preference</string>
+    <string name="pref_enable_datachannel_title">Enable datachannel.</string>
+    <string name="pref_enable_datachannel_default" translatable="false">true</string>
+
+    <string name="pref_ordered_key">ordered_preference</string>
+    <string name="pref_ordered_title">Order messages.</string>
+    <string name="pref_ordered_default" translatable="false">true</string>
+
+    <string name="pref_data_protocol_key">Subprotocol</string>
+    <string name="pref_data_protocol_title">Subprotocol.</string>
+    <string name="pref_data_protocol_dlg">Enter subprotocol.</string>
+    <string name="pref_data_protocol_default" translatable="false"></string>
+
+    <string name="pref_negotiated_key">negotiated_preference</string>
+    <string name="pref_negotiated_title">Negotiated.</string>
+    <string name="pref_negotiated_default" translatable="false">false</string>
+
+    <string name="pref_max_retransmit_time_ms_key">max_retransmit_time_ms_preference</string>
+    <string name="pref_max_retransmit_time_ms_title">Max delay to retransmit.</string>
+    <string name="pref_max_retransmit_time_ms_dlg">Enter max delay to retransmit (in ms).</string>
+    <string name="pref_max_retransmit_time_ms_default" translatable="false">-1</string>
+
+    <string name="pref_max_retransmits_key">max_retransmits_preference</string>
+    <string name="pref_max_retransmits_title">Max attempts to retransmit.</string>
+    <string name="pref_max_retransmits_dlg">Enter max attempts to retransmit.</string>
+    <string name="pref_max_retransmits_default" translatable="false">-1</string>
+
+    <string name="pref_data_id_key">data_id_preference</string>
+    <string name="pref_data_id_title">Data id.</string>
+    <string name="pref_data_id_dlg">Enter data channel id.</string>
+    <string name="pref_data_id_default" translatable="false">-1</string>
+
+    <string name="pref_miscsettings_key">misc_settings_key</string>
+    <string name="pref_miscsettings_title">Miscellaneous settings.</string>
+
+    <string name="pref_room_server_url_key">room_server_url_preference</string>
+    <string name="pref_room_server_url_title">Room server URL.</string>
+    <string name="pref_room_server_url_dlg">Enter a room server URL.</string>
+    <string name="pref_room_server_url_default" translatable="false">https://appr.tc</string>
+
+    <string name="pref_displayhud_key">displayhud_preference</string>
+    <string name="pref_displayhud_title">Display call statistics.</string>
+    <string name="pref_displayhud_dlg">Display call statistics.</string>
+    <string name="pref_displayhud_default" translatable="false">false</string>
+
+    <string name="pref_tracing_key">tracing_preference</string>
+    <string name="pref_tracing_title">Debug performance tracing.</string>
+    <string name="pref_tracing_dlg">Debug performance tracing.</string>
+    <string name="pref_tracing_default" translatable="false">false</string>
+
+    <string name="pref_enable_rtceventlog_key">enable_rtceventlog_key</string>
+    <string name="pref_enable_rtceventlog_title">Enable RtcEventLog.</string>
+    <string name="pref_enable_rtceventlog_default">false</string>
+</resources>
diff --git a/third_party/libwebrtc/examples/androidapp/res/xml/preferences.xml b/third_party/libwebrtc/examples/androidapp/res/xml/preferences.xml
new file mode 100644
index 0000000000..14e74d5c0b
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/res/xml/preferences.xml
@@ -0,0 +1,247 @@
+<?xml version="1.0" encoding="utf-8"?>
+<PreferenceScreen xmlns:android="http://schemas.android.com/apk/res/android">
+    <PreferenceCategory
+        android:key="@string/pref_videosettings_key"
+        android:title="@string/pref_videosettings_title">
+
+        <CheckBoxPreference
+            android:key="@string/pref_videocall_key"
+            android:title="@string/pref_videocall_title"
+            android:dialogTitle="@string/pref_videocall_dlg"
+            android:defaultValue="@string/pref_videocall_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_screencapture_key"
+            android:title="@string/pref_screencapture_title"
+            android:defaultValue="@string/pref_screencapture_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_camera2_key"
+            android:title="@string/pref_camera2_title"
+            android:defaultValue="@string/pref_camera2_default" />
+
+        <ListPreference
+            android:key="@string/pref_resolution_key"
+            android:title="@string/pref_resolution_title"
+            android:defaultValue="@string/pref_resolution_default"
+            android:dialogTitle="@string/pref_resolution_dlg"
+            android:entries="@array/videoResolutions"
+            android:entryValues="@array/videoResolutionsValues" />
+
+        <ListPreference
+            android:key="@string/pref_fps_key"
+            android:title="@string/pref_fps_title"
+            android:defaultValue="@string/pref_fps_default"
+            android:dialogTitle="@string/pref_fps_dlg"
+            android:entries="@array/cameraFps"
+            android:entryValues="@array/cameraFps" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_capturequalityslider_key"
+            android:title="@string/pref_capturequalityslider_title"
+            android:dialogTitle="@string/pref_capturequalityslider_dlg"
+            android:defaultValue="@string/pref_capturequalityslider_default" />
+
+        <ListPreference
+            android:key="@string/pref_maxvideobitrate_key"
+            android:title="@string/pref_maxvideobitrate_title"
+            android:defaultValue="@string/pref_maxvideobitrate_default"
+            android:dialogTitle="@string/pref_maxvideobitrate_dlg"
+            android:entries="@array/startBitrate"
+            android:entryValues="@array/startBitrate" />
+
+        <EditTextPreference
+            android:key="@string/pref_maxvideobitratevalue_key"
+            android:title="@string/pref_maxvideobitratevalue_title"
+            android:inputType="number"
+            android:defaultValue="@string/pref_maxvideobitratevalue_default"
+            android:dialogTitle="@string/pref_maxvideobitratevalue_dlg" />
+
+        <ListPreference
+            android:key="@string/pref_videocodec_key"
+            android:title="@string/pref_videocodec_title"
+            android:defaultValue="@string/pref_videocodec_default"
+            android:dialogTitle="@string/pref_videocodec_dlg"
+            android:entries="@array/videoCodecs"
+            android:entryValues="@array/videoCodecs" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_hwcodec_key"
+            android:title="@string/pref_hwcodec_title"
+            android:dialogTitle="@string/pref_hwcodec_dlg"
+            android:defaultValue="@string/pref_hwcodec_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_capturetotexture_key"
+            android:title="@string/pref_capturetotexture_title"
+            android:dialogTitle="@string/pref_capturetotexture_dlg"
+            android:defaultValue="@string/pref_capturetotexture_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_flexfec_key"
+            android:title="@string/pref_flexfec_title"
+            android:dialogTitle="@string/pref_flexfec_dlg"
+            android:defaultValue="@string/pref_flexfec_default" />
+    </PreferenceCategory>
+
+    <PreferenceCategory
+        android:key="@string/pref_audiosettings_key"
+        android:title="@string/pref_audiosettings_title">
+
+        <ListPreference
+            android:key="@string/pref_startaudiobitrate_key"
+            android:title="@string/pref_startaudiobitrate_title"
+            android:defaultValue="@string/pref_startaudiobitrate_default"
+            android:dialogTitle="@string/pref_startaudiobitrate_dlg"
+            android:entries="@array/startBitrate"
+            android:entryValues="@array/startBitrate" />
+
+        <EditTextPreference
+            android:key="@string/pref_startaudiobitratevalue_key"
+            android:title="@string/pref_startaudiobitratevalue_title"
+            android:inputType="number"
+            android:defaultValue="@string/pref_startaudiobitratevalue_default"
+            android:dialogTitle="@string/pref_startaudiobitratevalue_dlg" />
+
+        <ListPreference
+            android:key="@string/pref_audiocodec_key"
+            android:title="@string/pref_audiocodec_title"
+            android:defaultValue="@string/pref_audiocodec_default"
+            android:dialogTitle="@string/pref_audiocodec_dlg"
+            android:entries="@array/audioCodecs"
+            android:entryValues="@array/audioCodecs" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_noaudioprocessing_key"
+            android:title="@string/pref_noaudioprocessing_title"
+            android:dialogTitle="@string/pref_noaudioprocessing_dlg"
+            android:defaultValue="@string/pref_noaudioprocessing_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_aecdump_key"
+            android:title="@string/pref_aecdump_title"
+            android:dialogTitle="@string/pref_aecdump_dlg"
+            android:defaultValue="@string/pref_aecdump_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_enable_save_input_audio_to_file_key"
+            android:title="@string/pref_enable_save_input_audio_to_file_title"
+            android:dialogTitle="@string/pref_enable_save_input_audio_to_file_dlg"
+            android:defaultValue="@string/pref_enable_save_input_audio_to_file_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_opensles_key"
+            android:title="@string/pref_opensles_title"
+            android:dialogTitle="@string/pref_opensles_dlg"
+            android:defaultValue="@string/pref_opensles_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_disable_built_in_aec_key"
+            android:title="@string/pref_disable_built_in_aec_title"
+            android:dialogTitle="@string/pref_disable_built_in_aec_dlg"
+            android:defaultValue="@string/pref_disable_built_in_aec_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_disable_built_in_agc_key"
+            android:title="@string/pref_disable_built_in_agc_title"
+            android:dialogTitle="@string/pref_disable_built_in_agc_dlg"
+            android:defaultValue="@string/pref_disable_built_in_agc_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_disable_built_in_ns_key"
+            android:title="@string/pref_disable_built_in_ns_title"
+            android:dialogTitle="@string/pref_disable_built_in_ns_dlg"
+            android:defaultValue="@string/pref_disable_built_in_ns_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_disable_webrtc_agc_and_hpf_key"
+            android:title="@string/pref_disable_webrtc_agc_and_hpf_title"
+            android:defaultValue="@string/pref_disable_webrtc_agc_default" />
+
+        <ListPreference
+            android:key="@string/pref_speakerphone_key"
+            android:title="@string/pref_speakerphone_title"
+            android:defaultValue="@string/pref_speakerphone_default"
+            android:dialogTitle="@string/pref_speakerphone_dlg"
+            android:entries="@array/speakerphone"
+            android:entryValues="@array/speakerphoneValues" />
+    </PreferenceCategory>
+
+    <PreferenceCategory
+        android:key="@string/pref_datasettings_key"
+        android:title="@string/pref_datasettings_title">
+
+        <CheckBoxPreference
+            android:key="@string/pref_enable_datachannel_key"
+            android:title="@string/pref_enable_datachannel_title"
+            android:defaultValue="@string/pref_enable_datachannel_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_ordered_key"
+            android:title="@string/pref_ordered_title"
+            android:defaultValue="@string/pref_ordered_default" />
+
+        <EditTextPreference
+            android:key="@string/pref_data_protocol_key"
+            android:title="@string/pref_data_protocol_title"
+            android:inputType="text"
+            android:defaultValue="@string/pref_data_protocol_default"
+            android:dialogTitle="@string/pref_data_protocol_dlg" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_negotiated_key"
+            android:title="@string/pref_negotiated_title"
+            android:defaultValue="@string/pref_negotiated_default" />
+
+        <EditTextPreference
+            android:key="@string/pref_max_retransmit_time_ms_key"
+            android:title="@string/pref_max_retransmit_time_ms_title"
+            android:inputType="number"
+            android:defaultValue="@string/pref_max_retransmit_time_ms_default"
+            android:dialogTitle="@string/pref_max_retransmit_time_ms_dlg" />
+
+        <EditTextPreference
+            android:key="@string/pref_max_retransmits_key"
+            android:title="@string/pref_max_retransmits_title"
+            android:inputType="number"
+            android:defaultValue="@string/pref_max_retransmits_default"
+            android:dialogTitle="@string/pref_max_retransmits_dlg" />
+
+        <EditTextPreference
+            android:key="@string/pref_data_id_key"
+            android:title="@string/pref_data_id_title"
+            android:inputType="number"
+            android:defaultValue="@string/pref_data_id_default"
+            android:dialogTitle="@string/pref_data_id_dlg" />
+    </PreferenceCategory>
+
+    <PreferenceCategory
+        android:key="@string/pref_miscsettings_key"
+        android:title="@string/pref_miscsettings_title">
+
+        <EditTextPreference
+            android:key="@string/pref_room_server_url_key"
+            android:title="@string/pref_room_server_url_title"
+            android:inputType="text"
+            android:defaultValue="@string/pref_room_server_url_default"
+            android:dialogTitle="@string/pref_room_server_url_dlg" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_displayhud_key"
+            android:title="@string/pref_displayhud_title"
+            android:dialogTitle="@string/pref_displayhud_dlg"
+            android:defaultValue="@string/pref_displayhud_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_tracing_key"
+            android:title="@string/pref_tracing_title"
+            android:dialogTitle="@string/pref_tracing_dlg"
+            android:defaultValue="@string/pref_tracing_default" />
+
+        <CheckBoxPreference
+            android:key="@string/pref_enable_rtceventlog_key"
+            android:title="@string/pref_enable_rtceventlog_title"
+            android:defaultValue="@string/pref_enable_rtceventlog_default"/>
+    </PreferenceCategory>
+
+</PreferenceScreen>
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java
new file mode 100644
index 0000000000..2536b131a1
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java
@@ -0,0 +1,594 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.content.BroadcastReceiver;
+import android.content.Context;
+import android.content.Intent;
+import android.content.IntentFilter;
+import android.content.SharedPreferences;
+import android.content.pm.PackageManager;
+import android.media.AudioDeviceInfo;
+import android.media.AudioManager;
+import android.os.Build;
+import android.preference.PreferenceManager;
+import android.util.Log;
+import androidx.annotation.Nullable;
+import java.util.Collections;
+import java.util.HashSet;
+import java.util.Set;
+import org.appspot.apprtc.util.AppRTCUtils;
+import org.webrtc.ThreadUtils;
+
+/**
+ * AppRTCAudioManager manages all audio related parts of the AppRTC demo.
+ */
+public class AppRTCAudioManager {
+  private static final String TAG = "AppRTCAudioManager";
+  private static final String SPEAKERPHONE_AUTO = "auto";
+  private static final String SPEAKERPHONE_TRUE = "true";
+  private static final String SPEAKERPHONE_FALSE = "false";
+
+  /**
+   * AudioDevice is the names of possible audio devices that we currently
+   * support.
+   */
+  public enum AudioDevice { SPEAKER_PHONE, WIRED_HEADSET, EARPIECE, BLUETOOTH, NONE }
+
+  /** AudioManager state. */
+  public enum AudioManagerState {
+    UNINITIALIZED,
+    PREINITIALIZED,
+    RUNNING,
+  }
+
+  /** Selected audio device change event. */
+  public interface AudioManagerEvents {
+    // Callback fired once audio device is changed or list of available audio devices changed.
+    void onAudioDeviceChanged(
+        AudioDevice selectedAudioDevice, Set<AudioDevice> availableAudioDevices);
+  }
+
+  private final Context apprtcContext;
+  @Nullable
+  private AudioManager audioManager;
+
+  @Nullable
+  private AudioManagerEvents audioManagerEvents;
+  private AudioManagerState amState;
+  private int savedAudioMode = AudioManager.MODE_INVALID;
+  private boolean savedIsSpeakerPhoneOn;
+  private boolean savedIsMicrophoneMute;
+  private boolean hasWiredHeadset;
+
+  // Default audio device; speaker phone for video calls or earpiece for audio
+  // only calls.
+  private AudioDevice defaultAudioDevice;
+
+  // Contains the currently selected audio device.
+  // This device is changed automatically using a certain scheme where e.g.
+  // a wired headset "wins" over speaker phone. It is also possible for a
+  // user to explicitly select a device (and overrid any predefined scheme).
+  // See `userSelectedAudioDevice` for details.
+  private AudioDevice selectedAudioDevice;
+
+  // Contains the user-selected audio device which overrides the predefined
+  // selection scheme.
+  // TODO(henrika): always set to AudioDevice.NONE today. Add support for
+  // explicit selection based on choice by userSelectedAudioDevice.
+  private AudioDevice userSelectedAudioDevice;
+
+  // Contains speakerphone setting: auto, true or false
+  @Nullable private final String useSpeakerphone;
+
+  // Proximity sensor object. It measures the proximity of an object in cm
+  // relative to the view screen of a device and can therefore be used to
+  // assist device switching (close to ear <=> use headset earpiece if
+  // available, far from ear <=> use speaker phone).
+  @Nullable private AppRTCProximitySensor proximitySensor;
+
+  // Handles all tasks related to Bluetooth headset devices.
+  private final AppRTCBluetoothManager bluetoothManager;
+
+  // Contains a list of available audio devices. A Set collection is used to
+  // avoid duplicate elements.
+  private Set<AudioDevice> audioDevices = new HashSet<>();
+
+  // Broadcast receiver for wired headset intent broadcasts.
+  private BroadcastReceiver wiredHeadsetReceiver;
+
+  // Callback method for changes in audio focus.
+  @Nullable
+  private AudioManager.OnAudioFocusChangeListener audioFocusChangeListener;
+
+  /**
+   * This method is called when the proximity sensor reports a state change,
+   * e.g. from "NEAR to FAR" or from "FAR to NEAR".
+   */
+  private void onProximitySensorChangedState() {
+    if (!useSpeakerphone.equals(SPEAKERPHONE_AUTO)) {
+      return;
+    }
+
+    // The proximity sensor should only be activated when there are exactly two
+    // available audio devices.
+    if (audioDevices.size() == 2 && audioDevices.contains(AppRTCAudioManager.AudioDevice.EARPIECE)
+        && audioDevices.contains(AppRTCAudioManager.AudioDevice.SPEAKER_PHONE)) {
+      if (proximitySensor.sensorReportsNearState()) {
+        // Sensor reports that a "handset is being held up to a person's ear",
+        // or "something is covering the light sensor".
+        setAudioDeviceInternal(AppRTCAudioManager.AudioDevice.EARPIECE);
+      } else {
+        // Sensor reports that a "handset is removed from a person's ear", or
+        // "the light sensor is no longer covered".
+        setAudioDeviceInternal(AppRTCAudioManager.AudioDevice.SPEAKER_PHONE);
+      }
+    }
+  }
+
+  /* Receiver which handles changes in wired headset availability. */
+  private class WiredHeadsetReceiver extends BroadcastReceiver {
+    private static final int STATE_UNPLUGGED = 0;
+    private static final int STATE_PLUGGED = 1;
+    private static final int HAS_NO_MIC = 0;
+    private static final int HAS_MIC = 1;
+
+    @Override
+    public void onReceive(Context context, Intent intent) {
+      int state = intent.getIntExtra("state", STATE_UNPLUGGED);
+      int microphone = intent.getIntExtra("microphone", HAS_NO_MIC);
+      String name = intent.getStringExtra("name");
+      Log.d(TAG, "WiredHeadsetReceiver.onReceive" + AppRTCUtils.getThreadInfo() + ": "
+              + "a=" + intent.getAction() + ", s="
+              + (state == STATE_UNPLUGGED ? "unplugged" : "plugged") + ", m="
+              + (microphone == HAS_MIC ? "mic" : "no mic") + ", n=" + name + ", sb="
+              + isInitialStickyBroadcast());
+      hasWiredHeadset = (state == STATE_PLUGGED);
+      updateAudioDeviceState();
+    }
+  }
+
+  /** Construction. */
+  static AppRTCAudioManager create(Context context) {
+    return new AppRTCAudioManager(context);
+  }
+
+  private AppRTCAudioManager(Context context) {
+    Log.d(TAG, "ctor");
+    ThreadUtils.checkIsOnMainThread();
+    apprtcContext = context;
+    audioManager = ((AudioManager) context.getSystemService(Context.AUDIO_SERVICE));
+    bluetoothManager = AppRTCBluetoothManager.create(context, this);
+    wiredHeadsetReceiver = new WiredHeadsetReceiver();
+    amState = AudioManagerState.UNINITIALIZED;
+
+    SharedPreferences sharedPreferences = PreferenceManager.getDefaultSharedPreferences(context);
+    useSpeakerphone = sharedPreferences.getString(context.getString(R.string.pref_speakerphone_key),
+        context.getString(R.string.pref_speakerphone_default));
+    Log.d(TAG, "useSpeakerphone: " + useSpeakerphone);
+    if (useSpeakerphone.equals(SPEAKERPHONE_FALSE)) {
+      defaultAudioDevice = AudioDevice.EARPIECE;
+    } else {
+      defaultAudioDevice = AudioDevice.SPEAKER_PHONE;
+    }
+
+    // Create and initialize the proximity sensor.
+    // Tablet devices (e.g. Nexus 7) does not support proximity sensors.
+    // Note that, the sensor will not be active until start() has been called.
+    proximitySensor = AppRTCProximitySensor.create(context,
+        // This method will be called each time a state change is detected.
+        // Example: user holds their hand over the device (closer than ~5 cm),
+        // or removes their hand from the device.
+        this ::onProximitySensorChangedState);
+
+    Log.d(TAG, "defaultAudioDevice: " + defaultAudioDevice);
+    AppRTCUtils.logDeviceInfo(TAG);
+  }
+
+  @SuppressWarnings("deprecation") // TODO(henrika): audioManager.requestAudioFocus() is deprecated.
+  public void start(AudioManagerEvents audioManagerEvents) {
+    Log.d(TAG, "start");
+    ThreadUtils.checkIsOnMainThread();
+    if (amState == AudioManagerState.RUNNING) {
+      Log.e(TAG, "AudioManager is already active");
+      return;
+    }
+    // TODO(henrika): perhaps call new method called preInitAudio() here if UNINITIALIZED.
+
+    Log.d(TAG, "AudioManager starts...");
+    this.audioManagerEvents = audioManagerEvents;
+    amState = AudioManagerState.RUNNING;
+
+    // Store current audio state so we can restore it when stop() is called.
+    savedAudioMode = audioManager.getMode();
+    savedIsSpeakerPhoneOn = audioManager.isSpeakerphoneOn();
+    savedIsMicrophoneMute = audioManager.isMicrophoneMute();
+    hasWiredHeadset = hasWiredHeadset();
+
+    // Create an AudioManager.OnAudioFocusChangeListener instance.
+    audioFocusChangeListener = new AudioManager.OnAudioFocusChangeListener() {
+      // Called on the listener to notify if the audio focus for this listener has been changed.
+      // The `focusChange` value indicates whether the focus was gained, whether the focus was lost,
+      // and whether that loss is transient, or whether the new focus holder will hold it for an
+      // unknown amount of time.
+      // TODO(henrika): possibly extend support of handling audio-focus changes. Only contains
+      // logging for now.
+      @Override
+      public void onAudioFocusChange(int focusChange) {
+        final String typeOfChange;
+        switch (focusChange) {
+          case AudioManager.AUDIOFOCUS_GAIN:
+            typeOfChange = "AUDIOFOCUS_GAIN";
+            break;
+          case AudioManager.AUDIOFOCUS_GAIN_TRANSIENT:
+            typeOfChange = "AUDIOFOCUS_GAIN_TRANSIENT";
+            break;
+          case AudioManager.AUDIOFOCUS_GAIN_TRANSIENT_EXCLUSIVE:
+            typeOfChange = "AUDIOFOCUS_GAIN_TRANSIENT_EXCLUSIVE";
+            break;
+          case AudioManager.AUDIOFOCUS_GAIN_TRANSIENT_MAY_DUCK:
+            typeOfChange = "AUDIOFOCUS_GAIN_TRANSIENT_MAY_DUCK";
+            break;
+          case AudioManager.AUDIOFOCUS_LOSS:
+            typeOfChange = "AUDIOFOCUS_LOSS";
+            break;
+          case AudioManager.AUDIOFOCUS_LOSS_TRANSIENT:
+            typeOfChange = "AUDIOFOCUS_LOSS_TRANSIENT";
+            break;
+          case AudioManager.AUDIOFOCUS_LOSS_TRANSIENT_CAN_DUCK:
+            typeOfChange = "AUDIOFOCUS_LOSS_TRANSIENT_CAN_DUCK";
+            break;
+          default:
+            typeOfChange = "AUDIOFOCUS_INVALID";
+            break;
+        }
+        Log.d(TAG, "onAudioFocusChange: " + typeOfChange);
+      }
+    };
+
+    // Request audio playout focus (without ducking) and install listener for changes in focus.
+    int result = audioManager.requestAudioFocus(audioFocusChangeListener,
+        AudioManager.STREAM_VOICE_CALL, AudioManager.AUDIOFOCUS_GAIN_TRANSIENT);
+    if (result == AudioManager.AUDIOFOCUS_REQUEST_GRANTED) {
+      Log.d(TAG, "Audio focus request granted for VOICE_CALL streams");
+    } else {
+      Log.e(TAG, "Audio focus request failed");
+    }
+
+    // Start by setting MODE_IN_COMMUNICATION as default audio mode. It is
+    // required to be in this mode when playout and/or recording starts for
+    // best possible VoIP performance.
+    audioManager.setMode(AudioManager.MODE_IN_COMMUNICATION);
+
+    // Always disable microphone mute during a WebRTC call.
+    setMicrophoneMute(false);
+
+    // Set initial device states.
+    userSelectedAudioDevice = AudioDevice.NONE;
+    selectedAudioDevice = AudioDevice.NONE;
+    audioDevices.clear();
+
+    // Initialize and start Bluetooth if a BT device is available or initiate
+    // detection of new (enabled) BT devices.
+    bluetoothManager.start();
+
+    // Do initial selection of audio device. This setting can later be changed
+    // either by adding/removing a BT or wired headset or by covering/uncovering
+    // the proximity sensor.
+    updateAudioDeviceState();
+
+    // Register receiver for broadcast intents related to adding/removing a
+    // wired headset.
+    registerReceiver(wiredHeadsetReceiver, new IntentFilter(Intent.ACTION_HEADSET_PLUG));
+    Log.d(TAG, "AudioManager started");
+  }
+
+  @SuppressWarnings("deprecation") // TODO(henrika): audioManager.abandonAudioFocus() is deprecated.
+  public void stop() {
+    Log.d(TAG, "stop");
+    ThreadUtils.checkIsOnMainThread();
+    if (amState != AudioManagerState.RUNNING) {
+      Log.e(TAG, "Trying to stop AudioManager in incorrect state: " + amState);
+      return;
+    }
+    amState = AudioManagerState.UNINITIALIZED;
+
+    unregisterReceiver(wiredHeadsetReceiver);
+
+    bluetoothManager.stop();
+
+    // Restore previously stored audio states.
+    setSpeakerphoneOn(savedIsSpeakerPhoneOn);
+    setMicrophoneMute(savedIsMicrophoneMute);
+    audioManager.setMode(savedAudioMode);
+
+    // Abandon audio focus. Gives the previous focus owner, if any, focus.
+    audioManager.abandonAudioFocus(audioFocusChangeListener);
+    audioFocusChangeListener = null;
+    Log.d(TAG, "Abandoned audio focus for VOICE_CALL streams");
+
+    if (proximitySensor != null) {
+      proximitySensor.stop();
+      proximitySensor = null;
+    }
+
+    audioManagerEvents = null;
+    Log.d(TAG, "AudioManager stopped");
+  }
+
+  /** Changes selection of the currently active audio device. */
+  private void setAudioDeviceInternal(AudioDevice device) {
+    Log.d(TAG, "setAudioDeviceInternal(device=" + device + ")");
+    AppRTCUtils.assertIsTrue(audioDevices.contains(device));
+
+    switch (device) {
+      case SPEAKER_PHONE:
+        setSpeakerphoneOn(true);
+        break;
+      case EARPIECE:
+        setSpeakerphoneOn(false);
+        break;
+      case WIRED_HEADSET:
+        setSpeakerphoneOn(false);
+        break;
+      case BLUETOOTH:
+        setSpeakerphoneOn(false);
+        break;
+      default:
+        Log.e(TAG, "Invalid audio device selection");
+        break;
+    }
+    selectedAudioDevice = device;
+  }
+
+  /**
+   * Changes default audio device.
+   * TODO(henrika): add usage of this method in the AppRTCMobile client.
+   */
+  public void setDefaultAudioDevice(AudioDevice defaultDevice) {
+    ThreadUtils.checkIsOnMainThread();
+    switch (defaultDevice) {
+      case SPEAKER_PHONE:
+        defaultAudioDevice = defaultDevice;
+        break;
+      case EARPIECE:
+        if (hasEarpiece()) {
+          defaultAudioDevice = defaultDevice;
+        } else {
+          defaultAudioDevice = AudioDevice.SPEAKER_PHONE;
+        }
+        break;
+      default:
+        Log.e(TAG, "Invalid default audio device selection");
+        break;
+    }
+    Log.d(TAG, "setDefaultAudioDevice(device=" + defaultAudioDevice + ")");
+    updateAudioDeviceState();
+  }
+
+  /** Changes selection of the currently active audio device. */
+  public void selectAudioDevice(AudioDevice device) {
+    ThreadUtils.checkIsOnMainThread();
+    if (!audioDevices.contains(device)) {
+      Log.e(TAG, "Can not select " + device + " from available " + audioDevices);
+    }
+    userSelectedAudioDevice = device;
+    updateAudioDeviceState();
+  }
+
+  /** Returns current set of available/selectable audio devices. */
+  public Set<AudioDevice> getAudioDevices() {
+    ThreadUtils.checkIsOnMainThread();
+    return Collections.unmodifiableSet(new HashSet<>(audioDevices));
+  }
+
+  /** Returns the currently selected audio device. */
+  public AudioDevice getSelectedAudioDevice() {
+    ThreadUtils.checkIsOnMainThread();
+    return selectedAudioDevice;
+  }
+
+  /** Helper method for receiver registration. */
+  private void registerReceiver(BroadcastReceiver receiver, IntentFilter filter) {
+    apprtcContext.registerReceiver(receiver, filter);
+  }
+
+  /** Helper method for unregistration of an existing receiver. */
+  private void unregisterReceiver(BroadcastReceiver receiver) {
+    apprtcContext.unregisterReceiver(receiver);
+  }
+
+  /** Sets the speaker phone mode. */
+  private void setSpeakerphoneOn(boolean on) {
+    boolean wasOn = audioManager.isSpeakerphoneOn();
+    if (wasOn == on) {
+      return;
+    }
+    audioManager.setSpeakerphoneOn(on);
+  }
+
+  /** Sets the microphone mute state. */
+  private void setMicrophoneMute(boolean on) {
+    boolean wasMuted = audioManager.isMicrophoneMute();
+    if (wasMuted == on) {
+      return;
+    }
+    audioManager.setMicrophoneMute(on);
+  }
+
+  /** Gets the current earpiece state. */
+  private boolean hasEarpiece() {
+    return apprtcContext.getPackageManager().hasSystemFeature(PackageManager.FEATURE_TELEPHONY);
+  }
+
+  /**
+   * Checks whether a wired headset is connected or not.
+   * This is not a valid indication that audio playback is actually over
+   * the wired headset as audio routing depends on other conditions. We
+   * only use it as an early indicator (during initialization) of an attached
+   * wired headset.
+   */
+  @Deprecated
+  private boolean hasWiredHeadset() {
+    if (Build.VERSION.SDK_INT < Build.VERSION_CODES.M) {
+      return audioManager.isWiredHeadsetOn();
+    } else {
+      final AudioDeviceInfo[] devices = audioManager.getDevices(AudioManager.GET_DEVICES_ALL);
+      for (AudioDeviceInfo device : devices) {
+        final int type = device.getType();
+        if (type == AudioDeviceInfo.TYPE_WIRED_HEADSET) {
+          Log.d(TAG, "hasWiredHeadset: found wired headset");
+          return true;
+        } else if (type == AudioDeviceInfo.TYPE_USB_DEVICE) {
+          Log.d(TAG, "hasWiredHeadset: found USB audio device");
+          return true;
+        }
+      }
+      return false;
+    }
+  }
+
+  /**
+   * Updates list of possible audio devices and make new device selection.
+   * TODO(henrika): add unit test to verify all state transitions.
+   */
+  public void updateAudioDeviceState() {
+    ThreadUtils.checkIsOnMainThread();
+    Log.d(TAG, "--- updateAudioDeviceState: "
+            + "wired headset=" + hasWiredHeadset + ", "
+            + "BT state=" + bluetoothManager.getState());
+    Log.d(TAG, "Device status: "
+            + "available=" + audioDevices + ", "
+            + "selected=" + selectedAudioDevice + ", "
+            + "user selected=" + userSelectedAudioDevice);
+
+    // Check if any Bluetooth headset is connected. The internal BT state will
+    // change accordingly.
+    // TODO(henrika): perhaps wrap required state into BT manager.
+    if (bluetoothManager.getState() == AppRTCBluetoothManager.State.HEADSET_AVAILABLE
+        || bluetoothManager.getState() == AppRTCBluetoothManager.State.HEADSET_UNAVAILABLE
+        || bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_DISCONNECTING) {
+      bluetoothManager.updateDevice();
+    }
+
+    // Update the set of available audio devices.
+    Set<AudioDevice> newAudioDevices = new HashSet<>();
+
+    if (bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTED
+        || bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTING
+        || bluetoothManager.getState() == AppRTCBluetoothManager.State.HEADSET_AVAILABLE) {
+      newAudioDevices.add(AudioDevice.BLUETOOTH);
+    }
+
+    if (hasWiredHeadset) {
+      // If a wired headset is connected, then it is the only possible option.
+      newAudioDevices.add(AudioDevice.WIRED_HEADSET);
+    } else {
+      // No wired headset, hence the audio-device list can contain speaker
+      // phone (on a tablet), or speaker phone and earpiece (on mobile phone).
+      newAudioDevices.add(AudioDevice.SPEAKER_PHONE);
+      if (hasEarpiece()) {
+        newAudioDevices.add(AudioDevice.EARPIECE);
+      }
+    }
+    // Store state which is set to true if the device list has changed.
+    boolean audioDeviceSetUpdated = !audioDevices.equals(newAudioDevices);
+    // Update the existing audio device set.
+    audioDevices = newAudioDevices;
+    // Correct user selected audio devices if needed.
+    if (bluetoothManager.getState() == AppRTCBluetoothManager.State.HEADSET_UNAVAILABLE
+        && userSelectedAudioDevice == AudioDevice.BLUETOOTH) {
+      // If BT is not available, it can't be the user selection.
+      userSelectedAudioDevice = AudioDevice.NONE;
+    }
+    if (hasWiredHeadset && userSelectedAudioDevice == AudioDevice.SPEAKER_PHONE) {
+      // If user selected speaker phone, but then plugged wired headset then make
+      // wired headset as user selected device.
+      userSelectedAudioDevice = AudioDevice.WIRED_HEADSET;
+    }
+    if (!hasWiredHeadset && userSelectedAudioDevice == AudioDevice.WIRED_HEADSET) {
+      // If user selected wired headset, but then unplugged wired headset then make
+      // speaker phone as user selected device.
+      userSelectedAudioDevice = AudioDevice.SPEAKER_PHONE;
+    }
+
+    // Need to start Bluetooth if it is available and user either selected it explicitly or
+    // user did not select any output device.
+    boolean needBluetoothAudioStart =
+        bluetoothManager.getState() == AppRTCBluetoothManager.State.HEADSET_AVAILABLE
+        && (userSelectedAudioDevice == AudioDevice.NONE
+               || userSelectedAudioDevice == AudioDevice.BLUETOOTH);
+
+    // Need to stop Bluetooth audio if user selected different device and
+    // Bluetooth SCO connection is established or in the process.
+    boolean needBluetoothAudioStop =
+        (bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTED
+            || bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTING)
+        && (userSelectedAudioDevice != AudioDevice.NONE
+               && userSelectedAudioDevice != AudioDevice.BLUETOOTH);
+
+    if (bluetoothManager.getState() == AppRTCBluetoothManager.State.HEADSET_AVAILABLE
+        || bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTING
+        || bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTED) {
+      Log.d(TAG, "Need BT audio: start=" + needBluetoothAudioStart + ", "
+              + "stop=" + needBluetoothAudioStop + ", "
+              + "BT state=" + bluetoothManager.getState());
+    }
+
+    // Start or stop Bluetooth SCO connection given states set earlier.
+    if (needBluetoothAudioStop) {
+      bluetoothManager.stopScoAudio();
+      bluetoothManager.updateDevice();
+    }
+
+    if (needBluetoothAudioStart && !needBluetoothAudioStop) {
+      // Attempt to start Bluetooth SCO audio (takes a few second to start).
+      if (!bluetoothManager.startScoAudio()) {
+        // Remove BLUETOOTH from list of available devices since SCO failed.
+        audioDevices.remove(AudioDevice.BLUETOOTH);
+        audioDeviceSetUpdated = true;
+      }
+    }
+
+    // Update selected audio device.
+    final AudioDevice newAudioDevice;
+
+    if (bluetoothManager.getState() == AppRTCBluetoothManager.State.SCO_CONNECTED) {
+      // If a Bluetooth is connected, then it should be used as output audio
+      // device. Note that it is not sufficient that a headset is available;
+      // an active SCO channel must also be up and running.
+      newAudioDevice = AudioDevice.BLUETOOTH;
+    } else if (hasWiredHeadset) {
+      // If a wired headset is connected, but Bluetooth is not, then wired headset is used as
+      // audio device.
+      newAudioDevice = AudioDevice.WIRED_HEADSET;
+    } else {
+      // No wired headset and no Bluetooth, hence the audio-device list can contain speaker
+      // phone (on a tablet), or speaker phone and earpiece (on mobile phone).
+      // `defaultAudioDevice` contains either AudioDevice.SPEAKER_PHONE or AudioDevice.EARPIECE
+      // depending on the user's selection.
+      newAudioDevice = defaultAudioDevice;
+    }
+    // Switch to new device but only if there has been any changes.
+    if (newAudioDevice != selectedAudioDevice || audioDeviceSetUpdated) {
+      // Do the required device switch.
+      setAudioDeviceInternal(newAudioDevice);
+      Log.d(TAG, "New device status: "
+              + "available=" + audioDevices + ", "
+              + "selected=" + newAudioDevice);
+      if (audioManagerEvents != null) {
+        // Notify a listening client that audio device has been changed.
+        audioManagerEvents.onAudioDeviceChanged(selectedAudioDevice, audioDevices);
+      }
+    }
+    Log.d(TAG, "--- updateAudioDeviceState done");
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCBluetoothManager.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCBluetoothManager.java
new file mode 100644
index 0000000000..e9077d8bd6
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCBluetoothManager.java
@@ -0,0 +1,532 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.annotation.SuppressLint;
+import android.bluetooth.BluetoothAdapter;
+import android.bluetooth.BluetoothDevice;
+import android.bluetooth.BluetoothHeadset;
+import android.bluetooth.BluetoothProfile;
+import android.content.BroadcastReceiver;
+import android.content.Context;
+import android.content.Intent;
+import android.content.IntentFilter;
+import android.content.pm.PackageManager;
+import android.media.AudioManager;
+import android.os.Handler;
+import android.os.Looper;
+import android.os.Process;
+import android.util.Log;
+import androidx.annotation.Nullable;
+import java.util.List;
+import java.util.Set;
+import org.appspot.apprtc.util.AppRTCUtils;
+import org.webrtc.ThreadUtils;
+
+/**
+ * AppRTCProximitySensor manages functions related to Bluetoth devices in the
+ * AppRTC demo.
+ */
+public class AppRTCBluetoothManager {
+  private static final String TAG = "AppRTCBluetoothManager";
+
+  // Timeout interval for starting or stopping audio to a Bluetooth SCO device.
+  private static final int BLUETOOTH_SCO_TIMEOUT_MS = 4000;
+  // Maximum number of SCO connection attempts.
+  private static final int MAX_SCO_CONNECTION_ATTEMPTS = 2;
+
+  // Bluetooth connection state.
+  public enum State {
+    // Bluetooth is not available; no adapter or Bluetooth is off.
+    UNINITIALIZED,
+    // Bluetooth error happened when trying to start Bluetooth.
+    ERROR,
+    // Bluetooth proxy object for the Headset profile exists, but no connected headset devices,
+    // SCO is not started or disconnected.
+    HEADSET_UNAVAILABLE,
+    // Bluetooth proxy object for the Headset profile connected, connected Bluetooth headset
+    // present, but SCO is not started or disconnected.
+    HEADSET_AVAILABLE,
+    // Bluetooth audio SCO connection with remote device is closing.
+    SCO_DISCONNECTING,
+    // Bluetooth audio SCO connection with remote device is initiated.
+    SCO_CONNECTING,
+    // Bluetooth audio SCO connection with remote device is established.
+    SCO_CONNECTED
+  }
+
+  private final Context apprtcContext;
+  private final AppRTCAudioManager apprtcAudioManager;
+  @Nullable
+  private final AudioManager audioManager;
+  private final Handler handler;
+
+  int scoConnectionAttempts;
+  private State bluetoothState;
+  private final BluetoothProfile.ServiceListener bluetoothServiceListener;
+  @Nullable
+  private BluetoothAdapter bluetoothAdapter;
+  @Nullable
+  private BluetoothHeadset bluetoothHeadset;
+  @Nullable
+  private BluetoothDevice bluetoothDevice;
+  private final BroadcastReceiver bluetoothHeadsetReceiver;
+
+  // Runs when the Bluetooth timeout expires. We use that timeout after calling
+  // startScoAudio() or stopScoAudio() because we're not guaranteed to get a
+  // callback after those calls.
+  private final Runnable bluetoothTimeoutRunnable = new Runnable() {
+    @Override
+    public void run() {
+      bluetoothTimeout();
+    }
+  };
+
+  /**
+   * Implementation of an interface that notifies BluetoothProfile IPC clients when they have been
+   * connected to or disconnected from the service.
+   */
+  private class BluetoothServiceListener implements BluetoothProfile.ServiceListener {
+    @Override
+    // Called to notify the client when the proxy object has been connected to the service.
+    // Once we have the profile proxy object, we can use it to monitor the state of the
+    // connection and perform other operations that are relevant to the headset profile.
+    public void onServiceConnected(int profile, BluetoothProfile proxy) {
+      if (profile != BluetoothProfile.HEADSET || bluetoothState == State.UNINITIALIZED) {
+        return;
+      }
+      Log.d(TAG, "BluetoothServiceListener.onServiceConnected: BT state=" + bluetoothState);
+      // Android only supports one connected Bluetooth Headset at a time.
+      bluetoothHeadset = (BluetoothHeadset) proxy;
+      updateAudioDeviceState();
+      Log.d(TAG, "onServiceConnected done: BT state=" + bluetoothState);
+    }
+
+    @Override
+    /** Notifies the client when the proxy object has been disconnected from the service. */
+    public void onServiceDisconnected(int profile) {
+      if (profile != BluetoothProfile.HEADSET || bluetoothState == State.UNINITIALIZED) {
+        return;
+      }
+      Log.d(TAG, "BluetoothServiceListener.onServiceDisconnected: BT state=" + bluetoothState);
+      stopScoAudio();
+      bluetoothHeadset = null;
+      bluetoothDevice = null;
+      bluetoothState = State.HEADSET_UNAVAILABLE;
+      updateAudioDeviceState();
+      Log.d(TAG, "onServiceDisconnected done: BT state=" + bluetoothState);
+    }
+  }
+
+  // Intent broadcast receiver which handles changes in Bluetooth device availability.
+  // Detects headset changes and Bluetooth SCO state changes.
+  private class BluetoothHeadsetBroadcastReceiver extends BroadcastReceiver {
+    @Override
+    public void onReceive(Context context, Intent intent) {
+      if (bluetoothState == State.UNINITIALIZED) {
+        return;
+      }
+      final String action = intent.getAction();
+      // Change in connection state of the Headset profile. Note that the
+      // change does not tell us anything about whether we're streaming
+      // audio to BT over SCO. Typically received when user turns on a BT
+      // headset while audio is active using another audio device.
+      if (action.equals(BluetoothHeadset.ACTION_CONNECTION_STATE_CHANGED)) {
+        final int state =
+            intent.getIntExtra(BluetoothHeadset.EXTRA_STATE, BluetoothHeadset.STATE_DISCONNECTED);
+        Log.d(TAG, "BluetoothHeadsetBroadcastReceiver.onReceive: "
+                + "a=ACTION_CONNECTION_STATE_CHANGED, "
+                + "s=" + stateToString(state) + ", "
+                + "sb=" + isInitialStickyBroadcast() + ", "
+                + "BT state: " + bluetoothState);
+        if (state == BluetoothHeadset.STATE_CONNECTED) {
+          scoConnectionAttempts = 0;
+          updateAudioDeviceState();
+        } else if (state == BluetoothHeadset.STATE_CONNECTING) {
+          // No action needed.
+        } else if (state == BluetoothHeadset.STATE_DISCONNECTING) {
+          // No action needed.
+        } else if (state == BluetoothHeadset.STATE_DISCONNECTED) {
+          // Bluetooth is probably powered off during the call.
+          stopScoAudio();
+          updateAudioDeviceState();
+        }
+        // Change in the audio (SCO) connection state of the Headset profile.
+        // Typically received after call to startScoAudio() has finalized.
+      } else if (action.equals(BluetoothHeadset.ACTION_AUDIO_STATE_CHANGED)) {
+        final int state = intent.getIntExtra(
+            BluetoothHeadset.EXTRA_STATE, BluetoothHeadset.STATE_AUDIO_DISCONNECTED);
+        Log.d(TAG, "BluetoothHeadsetBroadcastReceiver.onReceive: "
+                + "a=ACTION_AUDIO_STATE_CHANGED, "
+                + "s=" + stateToString(state) + ", "
+                + "sb=" + isInitialStickyBroadcast() + ", "
+                + "BT state: " + bluetoothState);
+        if (state == BluetoothHeadset.STATE_AUDIO_CONNECTED) {
+          cancelTimer();
+          if (bluetoothState == State.SCO_CONNECTING) {
+            Log.d(TAG, "+++ Bluetooth audio SCO is now connected");
+            bluetoothState = State.SCO_CONNECTED;
+            scoConnectionAttempts = 0;
+            updateAudioDeviceState();
+          } else {
+            Log.w(TAG, "Unexpected state BluetoothHeadset.STATE_AUDIO_CONNECTED");
+          }
+        } else if (state == BluetoothHeadset.STATE_AUDIO_CONNECTING) {
+          Log.d(TAG, "+++ Bluetooth audio SCO is now connecting...");
+        } else if (state == BluetoothHeadset.STATE_AUDIO_DISCONNECTED) {
+          Log.d(TAG, "+++ Bluetooth audio SCO is now disconnected");
+          if (isInitialStickyBroadcast()) {
+            Log.d(TAG, "Ignore STATE_AUDIO_DISCONNECTED initial sticky broadcast.");
+            return;
+          }
+          updateAudioDeviceState();
+        }
+      }
+      Log.d(TAG, "onReceive done: BT state=" + bluetoothState);
+    }
+  }
+
+  /** Construction. */
+  static AppRTCBluetoothManager create(Context context, AppRTCAudioManager audioManager) {
+    Log.d(TAG, "create" + AppRTCUtils.getThreadInfo());
+    return new AppRTCBluetoothManager(context, audioManager);
+  }
+
+  protected AppRTCBluetoothManager(Context context, AppRTCAudioManager audioManager) {
+    Log.d(TAG, "ctor");
+    ThreadUtils.checkIsOnMainThread();
+    apprtcContext = context;
+    apprtcAudioManager = audioManager;
+    this.audioManager = getAudioManager(context);
+    bluetoothState = State.UNINITIALIZED;
+    bluetoothServiceListener = new BluetoothServiceListener();
+    bluetoothHeadsetReceiver = new BluetoothHeadsetBroadcastReceiver();
+    handler = new Handler(Looper.getMainLooper());
+  }
+
+  /** Returns the internal state. */
+  public State getState() {
+    ThreadUtils.checkIsOnMainThread();
+    return bluetoothState;
+  }
+
+  /**
+   * Activates components required to detect Bluetooth devices and to enable
+   * BT SCO (audio is routed via BT SCO) for the headset profile. The end
+   * state will be HEADSET_UNAVAILABLE but a state machine has started which
+   * will start a state change sequence where the final outcome depends on
+   * if/when the BT headset is enabled.
+   * Example of state change sequence when start() is called while BT device
+   * is connected and enabled:
+   *   UNINITIALIZED --> HEADSET_UNAVAILABLE --> HEADSET_AVAILABLE -->
+   *   SCO_CONNECTING --> SCO_CONNECTED <==> audio is now routed via BT SCO.
+   * Note that the AppRTCAudioManager is also involved in driving this state
+   * change.
+   */
+  public void start() {
+    ThreadUtils.checkIsOnMainThread();
+    Log.d(TAG, "start");
+    if (!hasPermission(apprtcContext, android.Manifest.permission.BLUETOOTH)) {
+      Log.w(TAG, "Process (pid=" + Process.myPid() + ") lacks BLUETOOTH permission");
+      return;
+    }
+    if (bluetoothState != State.UNINITIALIZED) {
+      Log.w(TAG, "Invalid BT state");
+      return;
+    }
+    bluetoothHeadset = null;
+    bluetoothDevice = null;
+    scoConnectionAttempts = 0;
+    // Get a handle to the default local Bluetooth adapter.
+    bluetoothAdapter = BluetoothAdapter.getDefaultAdapter();
+    if (bluetoothAdapter == null) {
+      Log.w(TAG, "Device does not support Bluetooth");
+      return;
+    }
+    // Ensure that the device supports use of BT SCO audio for off call use cases.
+    if (!audioManager.isBluetoothScoAvailableOffCall()) {
+      Log.e(TAG, "Bluetooth SCO audio is not available off call");
+      return;
+    }
+    logBluetoothAdapterInfo(bluetoothAdapter);
+    // Establish a connection to the HEADSET profile (includes both Bluetooth Headset and
+    // Hands-Free) proxy object and install a listener.
+    if (!getBluetoothProfileProxy(
+            apprtcContext, bluetoothServiceListener, BluetoothProfile.HEADSET)) {
+      Log.e(TAG, "BluetoothAdapter.getProfileProxy(HEADSET) failed");
+      return;
+    }
+    // Register receivers for BluetoothHeadset change notifications.
+    IntentFilter bluetoothHeadsetFilter = new IntentFilter();
+    // Register receiver for change in connection state of the Headset profile.
+    bluetoothHeadsetFilter.addAction(BluetoothHeadset.ACTION_CONNECTION_STATE_CHANGED);
+    // Register receiver for change in audio connection state of the Headset profile.
+    bluetoothHeadsetFilter.addAction(BluetoothHeadset.ACTION_AUDIO_STATE_CHANGED);
+    registerReceiver(bluetoothHeadsetReceiver, bluetoothHeadsetFilter);
+    Log.d(TAG, "HEADSET profile state: "
+            + stateToString(bluetoothAdapter.getProfileConnectionState(BluetoothProfile.HEADSET)));
+    Log.d(TAG, "Bluetooth proxy for headset profile has started");
+    bluetoothState = State.HEADSET_UNAVAILABLE;
+    Log.d(TAG, "start done: BT state=" + bluetoothState);
+  }
+
+  /** Stops and closes all components related to Bluetooth audio. */
+  public void stop() {
+    ThreadUtils.checkIsOnMainThread();
+    Log.d(TAG, "stop: BT state=" + bluetoothState);
+    if (bluetoothAdapter == null) {
+      return;
+    }
+    // Stop BT SCO connection with remote device if needed.
+    stopScoAudio();
+    // Close down remaining BT resources.
+    if (bluetoothState == State.UNINITIALIZED) {
+      return;
+    }
+    unregisterReceiver(bluetoothHeadsetReceiver);
+    cancelTimer();
+    if (bluetoothHeadset != null) {
+      bluetoothAdapter.closeProfileProxy(BluetoothProfile.HEADSET, bluetoothHeadset);
+      bluetoothHeadset = null;
+    }
+    bluetoothAdapter = null;
+    bluetoothDevice = null;
+    bluetoothState = State.UNINITIALIZED;
+    Log.d(TAG, "stop done: BT state=" + bluetoothState);
+  }
+
+  /**
+   * Starts Bluetooth SCO connection with remote device.
+   * Note that the phone application always has the priority on the usage of the SCO connection
+   * for telephony. If this method is called while the phone is in call it will be ignored.
+   * Similarly, if a call is received or sent while an application is using the SCO connection,
+   * the connection will be lost for the application and NOT returned automatically when the call
+   * ends. Also note that: up to and including API version JELLY_BEAN_MR1, this method initiates a
+   * virtual voice call to the Bluetooth headset. After API version JELLY_BEAN_MR2 only a raw SCO
+   * audio connection is established.
+   * TODO(henrika): should we add support for virtual voice call to BT headset also for JBMR2 and
+   * higher. It might be required to initiates a virtual voice call since many devices do not
+   * accept SCO audio without a "call".
+   */
+  public boolean startScoAudio() {
+    ThreadUtils.checkIsOnMainThread();
+    Log.d(TAG, "startSco: BT state=" + bluetoothState + ", "
+            + "attempts: " + scoConnectionAttempts + ", "
+            + "SCO is on: " + isScoOn());
+    if (scoConnectionAttempts >= MAX_SCO_CONNECTION_ATTEMPTS) {
+      Log.e(TAG, "BT SCO connection fails - no more attempts");
+      return false;
+    }
+    if (bluetoothState != State.HEADSET_AVAILABLE) {
+      Log.e(TAG, "BT SCO connection fails - no headset available");
+      return false;
+    }
+    // Start BT SCO channel and wait for ACTION_AUDIO_STATE_CHANGED.
+    Log.d(TAG, "Starting Bluetooth SCO and waits for ACTION_AUDIO_STATE_CHANGED...");
+    // The SCO connection establishment can take several seconds, hence we cannot rely on the
+    // connection to be available when the method returns but instead register to receive the
+    // intent ACTION_SCO_AUDIO_STATE_UPDATED and wait for the state to be SCO_AUDIO_STATE_CONNECTED.
+    bluetoothState = State.SCO_CONNECTING;
+    audioManager.startBluetoothSco();
+    audioManager.setBluetoothScoOn(true);
+    scoConnectionAttempts++;
+    startTimer();
+    Log.d(TAG, "startScoAudio done: BT state=" + bluetoothState + ", "
+            + "SCO is on: " + isScoOn());
+    return true;
+  }
+
+  /** Stops Bluetooth SCO connection with remote device. */
+  public void stopScoAudio() {
+    ThreadUtils.checkIsOnMainThread();
+    Log.d(TAG, "stopScoAudio: BT state=" + bluetoothState + ", "
+            + "SCO is on: " + isScoOn());
+    if (bluetoothState != State.SCO_CONNECTING && bluetoothState != State.SCO_CONNECTED) {
+      return;
+    }
+    cancelTimer();
+    audioManager.stopBluetoothSco();
+    audioManager.setBluetoothScoOn(false);
+    bluetoothState = State.SCO_DISCONNECTING;
+    Log.d(TAG, "stopScoAudio done: BT state=" + bluetoothState + ", "
+            + "SCO is on: " + isScoOn());
+  }
+
+  /**
+   * Use the BluetoothHeadset proxy object (controls the Bluetooth Headset
+   * Service via IPC) to update the list of connected devices for the HEADSET
+   * profile. The internal state will change to HEADSET_UNAVAILABLE or to
+   * HEADSET_AVAILABLE and `bluetoothDevice` will be mapped to the connected
+   * device if available.
+   */
+  public void updateDevice() {
+    if (bluetoothState == State.UNINITIALIZED || bluetoothHeadset == null) {
+      return;
+    }
+    Log.d(TAG, "updateDevice");
+    // Get connected devices for the headset profile. Returns the set of
+    // devices which are in state STATE_CONNECTED. The BluetoothDevice class
+    // is just a thin wrapper for a Bluetooth hardware address.
+    List<BluetoothDevice> devices = bluetoothHeadset.getConnectedDevices();
+    if (devices.isEmpty()) {
+      bluetoothDevice = null;
+      bluetoothState = State.HEADSET_UNAVAILABLE;
+      Log.d(TAG, "No connected bluetooth headset");
+    } else {
+      // Always use first device in list. Android only supports one device.
+      bluetoothDevice = devices.get(0);
+      bluetoothState = State.HEADSET_AVAILABLE;
+      Log.d(TAG, "Connected bluetooth headset: "
+              + "name=" + bluetoothDevice.getName() + ", "
+              + "state=" + stateToString(bluetoothHeadset.getConnectionState(bluetoothDevice))
+              + ", SCO audio=" + bluetoothHeadset.isAudioConnected(bluetoothDevice));
+    }
+    Log.d(TAG, "updateDevice done: BT state=" + bluetoothState);
+  }
+
+  /**
+   * Stubs for test mocks.
+   */
+  @Nullable
+  protected AudioManager getAudioManager(Context context) {
+    return (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
+  }
+
+  protected void registerReceiver(BroadcastReceiver receiver, IntentFilter filter) {
+    apprtcContext.registerReceiver(receiver, filter);
+  }
+
+  protected void unregisterReceiver(BroadcastReceiver receiver) {
+    apprtcContext.unregisterReceiver(receiver);
+  }
+
+  protected boolean getBluetoothProfileProxy(
+      Context context, BluetoothProfile.ServiceListener listener, int profile) {
+    return bluetoothAdapter.getProfileProxy(context, listener, profile);
+  }
+
+  protected boolean hasPermission(Context context, String permission) {
+    return apprtcContext.checkPermission(permission, Process.myPid(), Process.myUid())
+        == PackageManager.PERMISSION_GRANTED;
+  }
+
+  /** Logs the state of the local Bluetooth adapter. */
+  @SuppressLint("HardwareIds")
+  protected void logBluetoothAdapterInfo(BluetoothAdapter localAdapter) {
+    Log.d(TAG, "BluetoothAdapter: "
+            + "enabled=" + localAdapter.isEnabled() + ", "
+            + "state=" + stateToString(localAdapter.getState()) + ", "
+            + "name=" + localAdapter.getName() + ", "
+            + "address=" + localAdapter.getAddress());
+    // Log the set of BluetoothDevice objects that are bonded (paired) to the local adapter.
+    Set<BluetoothDevice> pairedDevices = localAdapter.getBondedDevices();
+    if (!pairedDevices.isEmpty()) {
+      Log.d(TAG, "paired devices:");
+      for (BluetoothDevice device : pairedDevices) {
+        Log.d(TAG, " name=" + device.getName() + ", address=" + device.getAddress());
+      }
+    }
+  }
+
+  /** Ensures that the audio manager updates its list of available audio devices. */
+  private void updateAudioDeviceState() {
+    ThreadUtils.checkIsOnMainThread();
+    Log.d(TAG, "updateAudioDeviceState");
+    apprtcAudioManager.updateAudioDeviceState();
+  }
+
+  /** Starts timer which times out after BLUETOOTH_SCO_TIMEOUT_MS milliseconds. */
+  private void startTimer() {
+    ThreadUtils.checkIsOnMainThread();
+    Log.d(TAG, "startTimer");
+    handler.postDelayed(bluetoothTimeoutRunnable, BLUETOOTH_SCO_TIMEOUT_MS);
+  }
+
+  /** Cancels any outstanding timer tasks. */
+  private void cancelTimer() {
+    ThreadUtils.checkIsOnMainThread();
+    Log.d(TAG, "cancelTimer");
+    handler.removeCallbacks(bluetoothTimeoutRunnable);
+  }
+
+  /**
+   * Called when start of the BT SCO channel takes too long time. Usually
+   * happens when the BT device has been turned on during an ongoing call.
+   */
+  private void bluetoothTimeout() {
+    ThreadUtils.checkIsOnMainThread();
+    if (bluetoothState == State.UNINITIALIZED || bluetoothHeadset == null) {
+      return;
+    }
+    Log.d(TAG, "bluetoothTimeout: BT state=" + bluetoothState + ", "
+            + "attempts: " + scoConnectionAttempts + ", "
+            + "SCO is on: " + isScoOn());
+    if (bluetoothState != State.SCO_CONNECTING) {
+      return;
+    }
+    // Bluetooth SCO should be connecting; check the latest result.
+    boolean scoConnected = false;
+    List<BluetoothDevice> devices = bluetoothHeadset.getConnectedDevices();
+    if (devices.size() > 0) {
+      bluetoothDevice = devices.get(0);
+      if (bluetoothHeadset.isAudioConnected(bluetoothDevice)) {
+        Log.d(TAG, "SCO connected with " + bluetoothDevice.getName());
+        scoConnected = true;
+      } else {
+        Log.d(TAG, "SCO is not connected with " + bluetoothDevice.getName());
+      }
+    }
+    if (scoConnected) {
+      // We thought BT had timed out, but it's actually on; updating state.
+      bluetoothState = State.SCO_CONNECTED;
+      scoConnectionAttempts = 0;
+    } else {
+      // Give up and "cancel" our request by calling stopBluetoothSco().
+      Log.w(TAG, "BT failed to connect after timeout");
+      stopScoAudio();
+    }
+    updateAudioDeviceState();
+    Log.d(TAG, "bluetoothTimeout done: BT state=" + bluetoothState);
+  }
+
+  /** Checks whether audio uses Bluetooth SCO. */
+  private boolean isScoOn() {
+    return audioManager.isBluetoothScoOn();
+  }
+
+  /** Converts BluetoothAdapter states into local string representations. */
+  private String stateToString(int state) {
+    switch (state) {
+      case BluetoothAdapter.STATE_DISCONNECTED:
+        return "DISCONNECTED";
+      case BluetoothAdapter.STATE_CONNECTED:
+        return "CONNECTED";
+      case BluetoothAdapter.STATE_CONNECTING:
+        return "CONNECTING";
+      case BluetoothAdapter.STATE_DISCONNECTING:
+        return "DISCONNECTING";
+      case BluetoothAdapter.STATE_OFF:
+        return "OFF";
+      case BluetoothAdapter.STATE_ON:
+        return "ON";
+      case BluetoothAdapter.STATE_TURNING_OFF:
+        // Indicates the local Bluetooth adapter is turning off. Local clients should immediately
+        // attempt graceful disconnection of any remote links.
+        return "TURNING_OFF";
+      case BluetoothAdapter.STATE_TURNING_ON:
+        // Indicates the local Bluetooth adapter is turning on. However local clients should wait
+        // for STATE_ON before attempting to use the adapter.
+        return  "TURNING_ON";
+      default:
+        return "INVALID";
+    }
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCClient.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCClient.java
new file mode 100644
index 0000000000..d5b7b4338e
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCClient.java
@@ -0,0 +1,137 @@
+/*
+ *  Copyright 2013 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import org.webrtc.IceCandidate;
+import org.webrtc.PeerConnection;
+import org.webrtc.SessionDescription;
+
+import java.util.List;
+
+/**
+ * AppRTCClient is the interface representing an AppRTC client.
+ */
+public interface AppRTCClient {
+  /**
+   * Struct holding the connection parameters of an AppRTC room.
+   */
+  class RoomConnectionParameters {
+    public final String roomUrl;
+    public final String roomId;
+    public final boolean loopback;
+    public final String urlParameters;
+    public RoomConnectionParameters(
+        String roomUrl, String roomId, boolean loopback, String urlParameters) {
+      this.roomUrl = roomUrl;
+      this.roomId = roomId;
+      this.loopback = loopback;
+      this.urlParameters = urlParameters;
+    }
+    public RoomConnectionParameters(String roomUrl, String roomId, boolean loopback) {
+      this(roomUrl, roomId, loopback, null /* urlParameters */);
+    }
+  }
+
+  /**
+   * Asynchronously connect to an AppRTC room URL using supplied connection
+   * parameters. Once connection is established onConnectedToRoom()
+   * callback with room parameters is invoked.
+   */
+  void connectToRoom(RoomConnectionParameters connectionParameters);
+
+  /**
+   * Send offer SDP to the other participant.
+   */
+  void sendOfferSdp(final SessionDescription sdp);
+
+  /**
+   * Send answer SDP to the other participant.
+   */
+  void sendAnswerSdp(final SessionDescription sdp);
+
+  /**
+   * Send Ice candidate to the other participant.
+   */
+  void sendLocalIceCandidate(final IceCandidate candidate);
+
+  /**
+   * Send removed ICE candidates to the other participant.
+   */
+  void sendLocalIceCandidateRemovals(final IceCandidate[] candidates);
+
+  /**
+   * Disconnect from room.
+   */
+  void disconnectFromRoom();
+
+  /**
+   * Struct holding the signaling parameters of an AppRTC room.
+   */
+  class SignalingParameters {
+    public final List<PeerConnection.IceServer> iceServers;
+    public final boolean initiator;
+    public final String clientId;
+    public final String wssUrl;
+    public final String wssPostUrl;
+    public final SessionDescription offerSdp;
+    public final List<IceCandidate> iceCandidates;
+
+    public SignalingParameters(List<PeerConnection.IceServer> iceServers, boolean initiator,
+        String clientId, String wssUrl, String wssPostUrl, SessionDescription offerSdp,
+        List<IceCandidate> iceCandidates) {
+      this.iceServers = iceServers;
+      this.initiator = initiator;
+      this.clientId = clientId;
+      this.wssUrl = wssUrl;
+      this.wssPostUrl = wssPostUrl;
+      this.offerSdp = offerSdp;
+      this.iceCandidates = iceCandidates;
+    }
+  }
+
+  /**
+   * Callback interface for messages delivered on signaling channel.
+   *
+   * <p>Methods are guaranteed to be invoked on the UI thread of `activity`.
+   */
+  interface SignalingEvents {
+    /**
+     * Callback fired once the room's signaling parameters
+     * SignalingParameters are extracted.
+     */
+    void onConnectedToRoom(final SignalingParameters params);
+
+    /**
+     * Callback fired once remote SDP is received.
+     */
+    void onRemoteDescription(final SessionDescription sdp);
+
+    /**
+     * Callback fired once remote Ice candidate is received.
+     */
+    void onRemoteIceCandidate(final IceCandidate candidate);
+
+    /**
+     * Callback fired once remote Ice candidate removals are received.
+     */
+    void onRemoteIceCandidatesRemoved(final IceCandidate[] candidates);
+
+    /**
+     * Callback fired once channel is closed.
+     */
+    void onChannelClose();
+
+    /**
+     * Callback fired once channel error happened.
+     */
+    void onChannelError(final String description);
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java
new file mode 100644
index 0000000000..604e2863d9
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java
@@ -0,0 +1,158 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.content.Context;
+import android.hardware.Sensor;
+import android.hardware.SensorEvent;
+import android.hardware.SensorEventListener;
+import android.hardware.SensorManager;
+import android.os.Build;
+import android.util.Log;
+import androidx.annotation.Nullable;
+import org.appspot.apprtc.util.AppRTCUtils;
+import org.webrtc.ThreadUtils;
+
+/**
+ * AppRTCProximitySensor manages functions related to the proximity sensor in
+ * the AppRTC demo.
+ * On most device, the proximity sensor is implemented as a boolean-sensor.
+ * It returns just two values "NEAR" or "FAR". Thresholding is done on the LUX
+ * value i.e. the LUX value of the light sensor is compared with a threshold.
+ * A LUX-value more than the threshold means the proximity sensor returns "FAR".
+ * Anything less than the threshold value and the sensor  returns "NEAR".
+ */
+public class AppRTCProximitySensor implements SensorEventListener {
+  private static final String TAG = "AppRTCProximitySensor";
+
+  // This class should be created, started and stopped on one thread
+  // (e.g. the main thread). We use `nonThreadSafe` to ensure that this is
+  // the case. Only active when `DEBUG` is set to true.
+  private final ThreadUtils.ThreadChecker threadChecker = new ThreadUtils.ThreadChecker();
+
+  private final Runnable onSensorStateListener;
+  private final SensorManager sensorManager;
+  @Nullable private Sensor proximitySensor;
+  private boolean lastStateReportIsNear;
+
+  /** Construction */
+  static AppRTCProximitySensor create(Context context, Runnable sensorStateListener) {
+    return new AppRTCProximitySensor(context, sensorStateListener);
+  }
+
+  private AppRTCProximitySensor(Context context, Runnable sensorStateListener) {
+    Log.d(TAG, "AppRTCProximitySensor" + AppRTCUtils.getThreadInfo());
+    onSensorStateListener = sensorStateListener;
+    sensorManager = ((SensorManager) context.getSystemService(Context.SENSOR_SERVICE));
+  }
+
+  /**
+   * Activate the proximity sensor. Also do initialization if called for the
+   * first time.
+   */
+  public boolean start() {
+    threadChecker.checkIsOnValidThread();
+    Log.d(TAG, "start" + AppRTCUtils.getThreadInfo());
+    if (!initDefaultSensor()) {
+      // Proximity sensor is not supported on this device.
+      return false;
+    }
+    sensorManager.registerListener(this, proximitySensor, SensorManager.SENSOR_DELAY_NORMAL);
+    return true;
+  }
+
+  /** Deactivate the proximity sensor. */
+  public void stop() {
+    threadChecker.checkIsOnValidThread();
+    Log.d(TAG, "stop" + AppRTCUtils.getThreadInfo());
+    if (proximitySensor == null) {
+      return;
+    }
+    sensorManager.unregisterListener(this, proximitySensor);
+  }
+
+  /** Getter for last reported state. Set to true if "near" is reported. */
+  public boolean sensorReportsNearState() {
+    threadChecker.checkIsOnValidThread();
+    return lastStateReportIsNear;
+  }
+
+  @Override
+  public final void onAccuracyChanged(Sensor sensor, int accuracy) {
+    threadChecker.checkIsOnValidThread();
+    AppRTCUtils.assertIsTrue(sensor.getType() == Sensor.TYPE_PROXIMITY);
+    if (accuracy == SensorManager.SENSOR_STATUS_UNRELIABLE) {
+      Log.e(TAG, "The values returned by this sensor cannot be trusted");
+    }
+  }
+
+  @Override
+  public final void onSensorChanged(SensorEvent event) {
+    threadChecker.checkIsOnValidThread();
+    AppRTCUtils.assertIsTrue(event.sensor.getType() == Sensor.TYPE_PROXIMITY);
+    // As a best practice; do as little as possible within this method and
+    // avoid blocking.
+    float distanceInCentimeters = event.values[0];
+    if (distanceInCentimeters < proximitySensor.getMaximumRange()) {
+      Log.d(TAG, "Proximity sensor => NEAR state");
+      lastStateReportIsNear = true;
+    } else {
+      Log.d(TAG, "Proximity sensor => FAR state");
+      lastStateReportIsNear = false;
+    }
+
+    // Report about new state to listening client. Client can then call
+    // sensorReportsNearState() to query the current state (NEAR or FAR).
+    if (onSensorStateListener != null) {
+      onSensorStateListener.run();
+    }
+
+    Log.d(TAG, "onSensorChanged" + AppRTCUtils.getThreadInfo() + ": "
+            + "accuracy=" + event.accuracy + ", timestamp=" + event.timestamp + ", distance="
+            + event.values[0]);
+  }
+
+  /**
+   * Get default proximity sensor if it exists. Tablet devices (e.g. Nexus 7)
+   * does not support this type of sensor and false will be returned in such
+   * cases.
+   */
+  private boolean initDefaultSensor() {
+    if (proximitySensor != null) {
+      return true;
+    }
+    proximitySensor = sensorManager.getDefaultSensor(Sensor.TYPE_PROXIMITY);
+    if (proximitySensor == null) {
+      return false;
+    }
+    logProximitySensorInfo();
+    return true;
+  }
+
+  /** Helper method for logging information about the proximity sensor. */
+  private void logProximitySensorInfo() {
+    if (proximitySensor == null) {
+      return;
+    }
+    StringBuilder info = new StringBuilder("Proximity sensor: ");
+    info.append("name=").append(proximitySensor.getName());
+    info.append(", vendor: ").append(proximitySensor.getVendor());
+    info.append(", power: ").append(proximitySensor.getPower());
+    info.append(", resolution: ").append(proximitySensor.getResolution());
+    info.append(", max range: ").append(proximitySensor.getMaximumRange());
+    info.append(", min delay: ").append(proximitySensor.getMinDelay());
+    info.append(", type: ").append(proximitySensor.getStringType());
+    info.append(", max delay: ").append(proximitySensor.getMaxDelay());
+    info.append(", reporting mode: ").append(proximitySensor.getReportingMode());
+    info.append(", isWakeUpSensor: ").append(proximitySensor.isWakeUpSensor());
+    Log.d(TAG, info.toString());
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java
new file mode 100644
index 0000000000..eb5ee8289e
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java
@@ -0,0 +1,962 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.annotation.TargetApi;
+import android.app.Activity;
+import android.app.AlertDialog;
+import android.app.FragmentTransaction;
+import android.content.Context;
+import android.content.DialogInterface;
+import android.content.Intent;
+import android.content.pm.PackageManager;
+import android.media.projection.MediaProjection;
+import android.media.projection.MediaProjectionManager;
+import android.net.Uri;
+import android.os.Build;
+import android.os.Bundle;
+import android.os.Handler;
+import android.util.DisplayMetrics;
+import android.util.Log;
+import android.view.View;
+import android.view.Window;
+import android.view.WindowManager;
+import android.view.WindowManager.LayoutParams;
+import android.widget.Toast;
+import androidx.annotation.Nullable;
+import java.io.IOException;
+import java.lang.RuntimeException;
+import java.util.ArrayList;
+import java.util.List;
+import java.util.Set;
+import org.appspot.apprtc.AppRTCAudioManager.AudioDevice;
+import org.appspot.apprtc.AppRTCAudioManager.AudioManagerEvents;
+import org.appspot.apprtc.AppRTCClient.RoomConnectionParameters;
+import org.appspot.apprtc.AppRTCClient.SignalingParameters;
+import org.appspot.apprtc.PeerConnectionClient.DataChannelParameters;
+import org.appspot.apprtc.PeerConnectionClient.PeerConnectionParameters;
+import org.webrtc.Camera1Enumerator;
+import org.webrtc.Camera2Enumerator;
+import org.webrtc.CameraEnumerator;
+import org.webrtc.EglBase;
+import org.webrtc.FileVideoCapturer;
+import org.webrtc.IceCandidate;
+import org.webrtc.Logging;
+import org.webrtc.PeerConnectionFactory;
+import org.webrtc.RTCStatsReport;
+import org.webrtc.RendererCommon.ScalingType;
+import org.webrtc.ScreenCapturerAndroid;
+import org.webrtc.SessionDescription;
+import org.webrtc.SurfaceViewRenderer;
+import org.webrtc.VideoCapturer;
+import org.webrtc.VideoFileRenderer;
+import org.webrtc.VideoFrame;
+import org.webrtc.VideoSink;
+
+/**
+ * Activity for peer connection call setup, call waiting
+ * and call view.
+ */
+public class CallActivity extends Activity implements AppRTCClient.SignalingEvents,
+                                                      PeerConnectionClient.PeerConnectionEvents,
+                                                      CallFragment.OnCallEvents {
+  private static final String TAG = "CallRTCClient";
+
+  public static final String EXTRA_ROOMID = "org.appspot.apprtc.ROOMID";
+  public static final String EXTRA_URLPARAMETERS = "org.appspot.apprtc.URLPARAMETERS";
+  public static final String EXTRA_LOOPBACK = "org.appspot.apprtc.LOOPBACK";
+  public static final String EXTRA_VIDEO_CALL = "org.appspot.apprtc.VIDEO_CALL";
+  public static final String EXTRA_SCREENCAPTURE = "org.appspot.apprtc.SCREENCAPTURE";
+  public static final String EXTRA_CAMERA2 = "org.appspot.apprtc.CAMERA2";
+  public static final String EXTRA_VIDEO_WIDTH = "org.appspot.apprtc.VIDEO_WIDTH";
+  public static final String EXTRA_VIDEO_HEIGHT = "org.appspot.apprtc.VIDEO_HEIGHT";
+  public static final String EXTRA_VIDEO_FPS = "org.appspot.apprtc.VIDEO_FPS";
+  public static final String EXTRA_VIDEO_CAPTUREQUALITYSLIDER_ENABLED =
+      "org.appsopt.apprtc.VIDEO_CAPTUREQUALITYSLIDER";
+  public static final String EXTRA_VIDEO_BITRATE = "org.appspot.apprtc.VIDEO_BITRATE";
+  public static final String EXTRA_VIDEOCODEC = "org.appspot.apprtc.VIDEOCODEC";
+  public static final String EXTRA_HWCODEC_ENABLED = "org.appspot.apprtc.HWCODEC";
+  public static final String EXTRA_CAPTURETOTEXTURE_ENABLED = "org.appspot.apprtc.CAPTURETOTEXTURE";
+  public static final String EXTRA_FLEXFEC_ENABLED = "org.appspot.apprtc.FLEXFEC";
+  public static final String EXTRA_AUDIO_BITRATE = "org.appspot.apprtc.AUDIO_BITRATE";
+  public static final String EXTRA_AUDIOCODEC = "org.appspot.apprtc.AUDIOCODEC";
+  public static final String EXTRA_NOAUDIOPROCESSING_ENABLED =
+      "org.appspot.apprtc.NOAUDIOPROCESSING";
+  public static final String EXTRA_AECDUMP_ENABLED = "org.appspot.apprtc.AECDUMP";
+  public static final String EXTRA_SAVE_INPUT_AUDIO_TO_FILE_ENABLED =
+      "org.appspot.apprtc.SAVE_INPUT_AUDIO_TO_FILE";
+  public static final String EXTRA_OPENSLES_ENABLED = "org.appspot.apprtc.OPENSLES";
+  public static final String EXTRA_DISABLE_BUILT_IN_AEC = "org.appspot.apprtc.DISABLE_BUILT_IN_AEC";
+  public static final String EXTRA_DISABLE_BUILT_IN_AGC = "org.appspot.apprtc.DISABLE_BUILT_IN_AGC";
+  public static final String EXTRA_DISABLE_BUILT_IN_NS = "org.appspot.apprtc.DISABLE_BUILT_IN_NS";
+  public static final String EXTRA_DISABLE_WEBRTC_AGC_AND_HPF =
+      "org.appspot.apprtc.DISABLE_WEBRTC_GAIN_CONTROL";
+  public static final String EXTRA_DISPLAY_HUD = "org.appspot.apprtc.DISPLAY_HUD";
+  public static final String EXTRA_TRACING = "org.appspot.apprtc.TRACING";
+  public static final String EXTRA_CMDLINE = "org.appspot.apprtc.CMDLINE";
+  public static final String EXTRA_RUNTIME = "org.appspot.apprtc.RUNTIME";
+  public static final String EXTRA_VIDEO_FILE_AS_CAMERA = "org.appspot.apprtc.VIDEO_FILE_AS_CAMERA";
+  public static final String EXTRA_SAVE_REMOTE_VIDEO_TO_FILE =
+      "org.appspot.apprtc.SAVE_REMOTE_VIDEO_TO_FILE";
+  public static final String EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_WIDTH =
+      "org.appspot.apprtc.SAVE_REMOTE_VIDEO_TO_FILE_WIDTH";
+  public static final String EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT =
+      "org.appspot.apprtc.SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT";
+  public static final String EXTRA_USE_VALUES_FROM_INTENT =
+      "org.appspot.apprtc.USE_VALUES_FROM_INTENT";
+  public static final String EXTRA_DATA_CHANNEL_ENABLED = "org.appspot.apprtc.DATA_CHANNEL_ENABLED";
+  public static final String EXTRA_ORDERED = "org.appspot.apprtc.ORDERED";
+  public static final String EXTRA_MAX_RETRANSMITS_MS = "org.appspot.apprtc.MAX_RETRANSMITS_MS";
+  public static final String EXTRA_MAX_RETRANSMITS = "org.appspot.apprtc.MAX_RETRANSMITS";
+  public static final String EXTRA_PROTOCOL = "org.appspot.apprtc.PROTOCOL";
+  public static final String EXTRA_NEGOTIATED = "org.appspot.apprtc.NEGOTIATED";
+  public static final String EXTRA_ID = "org.appspot.apprtc.ID";
+  public static final String EXTRA_ENABLE_RTCEVENTLOG = "org.appspot.apprtc.ENABLE_RTCEVENTLOG";
+
+  private static final int CAPTURE_PERMISSION_REQUEST_CODE = 1;
+
+  // List of mandatory application permissions.
+  private static final String[] MANDATORY_PERMISSIONS = {"android.permission.MODIFY_AUDIO_SETTINGS",
+      "android.permission.RECORD_AUDIO", "android.permission.INTERNET"};
+
+  // Peer connection statistics callback period in ms.
+  private static final int STAT_CALLBACK_PERIOD = 1000;
+
+  private static class ProxyVideoSink implements VideoSink {
+    private VideoSink target;
+
+    @Override
+    synchronized public void onFrame(VideoFrame frame) {
+      if (target == null) {
+        Logging.d(TAG, "Dropping frame in proxy because target is null.");
+        return;
+      }
+
+      target.onFrame(frame);
+    }
+
+    synchronized public void setTarget(VideoSink target) {
+      this.target = target;
+    }
+  }
+
+  private final ProxyVideoSink remoteProxyRenderer = new ProxyVideoSink();
+  private final ProxyVideoSink localProxyVideoSink = new ProxyVideoSink();
+  @Nullable private PeerConnectionClient peerConnectionClient;
+  @Nullable
+  private AppRTCClient appRtcClient;
+  @Nullable
+  private SignalingParameters signalingParameters;
+  @Nullable private AppRTCAudioManager audioManager;
+  @Nullable
+  private SurfaceViewRenderer pipRenderer;
+  @Nullable
+  private SurfaceViewRenderer fullscreenRenderer;
+  @Nullable
+  private VideoFileRenderer videoFileRenderer;
+  private final List<VideoSink> remoteSinks = new ArrayList<>();
+  private Toast logToast;
+  private boolean commandLineRun;
+  private boolean activityRunning;
+  private RoomConnectionParameters roomConnectionParameters;
+  @Nullable
+  private PeerConnectionParameters peerConnectionParameters;
+  private boolean connected;
+  private boolean isError;
+  private boolean callControlFragmentVisible = true;
+  private long callStartedTimeMs;
+  private boolean micEnabled = true;
+  private boolean screencaptureEnabled;
+  private static Intent mediaProjectionPermissionResultData;
+  private static int mediaProjectionPermissionResultCode;
+  // True if local view is in the fullscreen renderer.
+  private boolean isSwappedFeeds;
+
+  // Controls
+  private CallFragment callFragment;
+  private HudFragment hudFragment;
+  private CpuMonitor cpuMonitor;
+
+  @Override
+  // TODO(bugs.webrtc.org/8580): LayoutParams.FLAG_TURN_SCREEN_ON and
+  // LayoutParams.FLAG_SHOW_WHEN_LOCKED are deprecated.
+  @SuppressWarnings("deprecation")
+  public void onCreate(Bundle savedInstanceState) {
+    super.onCreate(savedInstanceState);
+    Thread.setDefaultUncaughtExceptionHandler(new UnhandledExceptionHandler(this));
+
+    // Set window styles for fullscreen-window size. Needs to be done before
+    // adding content.
+    requestWindowFeature(Window.FEATURE_NO_TITLE);
+    getWindow().addFlags(LayoutParams.FLAG_FULLSCREEN | LayoutParams.FLAG_KEEP_SCREEN_ON
+        | LayoutParams.FLAG_SHOW_WHEN_LOCKED | LayoutParams.FLAG_TURN_SCREEN_ON);
+    getWindow().getDecorView().setSystemUiVisibility(getSystemUiVisibility());
+    setContentView(R.layout.activity_call);
+
+    connected = false;
+    signalingParameters = null;
+
+    // Create UI controls.
+    pipRenderer = findViewById(R.id.pip_video_view);
+    fullscreenRenderer = findViewById(R.id.fullscreen_video_view);
+    callFragment = new CallFragment();
+    hudFragment = new HudFragment();
+
+    // Show/hide call control fragment on view click.
+    View.OnClickListener listener = new View.OnClickListener() {
+      @Override
+      public void onClick(View view) {
+        toggleCallControlFragmentVisibility();
+      }
+    };
+
+    // Swap feeds on pip view click.
+    pipRenderer.setOnClickListener(new View.OnClickListener() {
+      @Override
+      public void onClick(View view) {
+        setSwappedFeeds(!isSwappedFeeds);
+      }
+    });
+
+    fullscreenRenderer.setOnClickListener(listener);
+    remoteSinks.add(remoteProxyRenderer);
+
+    final Intent intent = getIntent();
+    final EglBase eglBase = EglBase.create();
+
+    // Create video renderers.
+    pipRenderer.init(eglBase.getEglBaseContext(), null);
+    pipRenderer.setScalingType(ScalingType.SCALE_ASPECT_FIT);
+    String saveRemoteVideoToFile = intent.getStringExtra(EXTRA_SAVE_REMOTE_VIDEO_TO_FILE);
+
+    // When saveRemoteVideoToFile is set we save the video from the remote to a file.
+    if (saveRemoteVideoToFile != null) {
+      int videoOutWidth = intent.getIntExtra(EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_WIDTH, 0);
+      int videoOutHeight = intent.getIntExtra(EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT, 0);
+      try {
+        videoFileRenderer = new VideoFileRenderer(
+            saveRemoteVideoToFile, videoOutWidth, videoOutHeight, eglBase.getEglBaseContext());
+        remoteSinks.add(videoFileRenderer);
+      } catch (IOException e) {
+        throw new RuntimeException(
+            "Failed to open video file for output: " + saveRemoteVideoToFile, e);
+      }
+    }
+    fullscreenRenderer.init(eglBase.getEglBaseContext(), null);
+    fullscreenRenderer.setScalingType(ScalingType.SCALE_ASPECT_FILL);
+
+    pipRenderer.setZOrderMediaOverlay(true);
+    pipRenderer.setEnableHardwareScaler(true /* enabled */);
+    fullscreenRenderer.setEnableHardwareScaler(false /* enabled */);
+    // Start with local feed in fullscreen and swap it to the pip when the call is connected.
+    setSwappedFeeds(true /* isSwappedFeeds */);
+
+    // Check for mandatory permissions.
+    for (String permission : MANDATORY_PERMISSIONS) {
+      if (checkCallingOrSelfPermission(permission) != PackageManager.PERMISSION_GRANTED) {
+        logAndToast("Permission " + permission + " is not granted");
+        setResult(RESULT_CANCELED);
+        finish();
+        return;
+      }
+    }
+
+    Uri roomUri = intent.getData();
+    if (roomUri == null) {
+      logAndToast(getString(R.string.missing_url));
+      Log.e(TAG, "Didn't get any URL in intent!");
+      setResult(RESULT_CANCELED);
+      finish();
+      return;
+    }
+
+    // Get Intent parameters.
+    String roomId = intent.getStringExtra(EXTRA_ROOMID);
+    Log.d(TAG, "Room ID: " + roomId);
+    if (roomId == null || roomId.length() == 0) {
+      logAndToast(getString(R.string.missing_url));
+      Log.e(TAG, "Incorrect room ID in intent!");
+      setResult(RESULT_CANCELED);
+      finish();
+      return;
+    }
+
+    boolean loopback = intent.getBooleanExtra(EXTRA_LOOPBACK, false);
+    boolean tracing = intent.getBooleanExtra(EXTRA_TRACING, false);
+
+    int videoWidth = intent.getIntExtra(EXTRA_VIDEO_WIDTH, 0);
+    int videoHeight = intent.getIntExtra(EXTRA_VIDEO_HEIGHT, 0);
+
+    screencaptureEnabled = intent.getBooleanExtra(EXTRA_SCREENCAPTURE, false);
+    // If capturing format is not specified for screencapture, use screen resolution.
+    if (screencaptureEnabled && videoWidth == 0 && videoHeight == 0) {
+      DisplayMetrics displayMetrics = getDisplayMetrics();
+      videoWidth = displayMetrics.widthPixels;
+      videoHeight = displayMetrics.heightPixels;
+    }
+    DataChannelParameters dataChannelParameters = null;
+    if (intent.getBooleanExtra(EXTRA_DATA_CHANNEL_ENABLED, false)) {
+      dataChannelParameters = new DataChannelParameters(intent.getBooleanExtra(EXTRA_ORDERED, true),
+          intent.getIntExtra(EXTRA_MAX_RETRANSMITS_MS, -1),
+          intent.getIntExtra(EXTRA_MAX_RETRANSMITS, -1), intent.getStringExtra(EXTRA_PROTOCOL),
+          intent.getBooleanExtra(EXTRA_NEGOTIATED, false), intent.getIntExtra(EXTRA_ID, -1));
+    }
+    peerConnectionParameters =
+        new PeerConnectionParameters(intent.getBooleanExtra(EXTRA_VIDEO_CALL, true), loopback,
+            tracing, videoWidth, videoHeight, intent.getIntExtra(EXTRA_VIDEO_FPS, 0),
+            intent.getIntExtra(EXTRA_VIDEO_BITRATE, 0), intent.getStringExtra(EXTRA_VIDEOCODEC),
+            intent.getBooleanExtra(EXTRA_HWCODEC_ENABLED, true),
+            intent.getBooleanExtra(EXTRA_FLEXFEC_ENABLED, false),
+            intent.getIntExtra(EXTRA_AUDIO_BITRATE, 0), intent.getStringExtra(EXTRA_AUDIOCODEC),
+            intent.getBooleanExtra(EXTRA_NOAUDIOPROCESSING_ENABLED, false),
+            intent.getBooleanExtra(EXTRA_AECDUMP_ENABLED, false),
+            intent.getBooleanExtra(EXTRA_SAVE_INPUT_AUDIO_TO_FILE_ENABLED, false),
+            intent.getBooleanExtra(EXTRA_OPENSLES_ENABLED, false),
+            intent.getBooleanExtra(EXTRA_DISABLE_BUILT_IN_AEC, false),
+            intent.getBooleanExtra(EXTRA_DISABLE_BUILT_IN_AGC, false),
+            intent.getBooleanExtra(EXTRA_DISABLE_BUILT_IN_NS, false),
+            intent.getBooleanExtra(EXTRA_DISABLE_WEBRTC_AGC_AND_HPF, false),
+            intent.getBooleanExtra(EXTRA_ENABLE_RTCEVENTLOG, false), dataChannelParameters);
+    commandLineRun = intent.getBooleanExtra(EXTRA_CMDLINE, false);
+    int runTimeMs = intent.getIntExtra(EXTRA_RUNTIME, 0);
+
+    Log.d(TAG, "VIDEO_FILE: '" + intent.getStringExtra(EXTRA_VIDEO_FILE_AS_CAMERA) + "'");
+
+    // Create connection client. Use DirectRTCClient if room name is an IP otherwise use the
+    // standard WebSocketRTCClient.
+    if (loopback || !DirectRTCClient.IP_PATTERN.matcher(roomId).matches()) {
+      appRtcClient = new WebSocketRTCClient(this);
+    } else {
+      Log.i(TAG, "Using DirectRTCClient because room name looks like an IP.");
+      appRtcClient = new DirectRTCClient(this);
+    }
+    // Create connection parameters.
+    String urlParameters = intent.getStringExtra(EXTRA_URLPARAMETERS);
+    roomConnectionParameters =
+        new RoomConnectionParameters(roomUri.toString(), roomId, loopback, urlParameters);
+
+    // Create CPU monitor
+    if (CpuMonitor.isSupported()) {
+      cpuMonitor = new CpuMonitor(this);
+      hudFragment.setCpuMonitor(cpuMonitor);
+    }
+
+    // Send intent arguments to fragments.
+    callFragment.setArguments(intent.getExtras());
+    hudFragment.setArguments(intent.getExtras());
+    // Activate call and HUD fragments and start the call.
+    FragmentTransaction ft = getFragmentManager().beginTransaction();
+    ft.add(R.id.call_fragment_container, callFragment);
+    ft.add(R.id.hud_fragment_container, hudFragment);
+    ft.commit();
+
+    // For command line execution run connection for <runTimeMs> and exit.
+    if (commandLineRun && runTimeMs > 0) {
+      (new Handler()).postDelayed(new Runnable() {
+        @Override
+        public void run() {
+          disconnect();
+        }
+      }, runTimeMs);
+    }
+
+    // Create peer connection client.
+    peerConnectionClient = new PeerConnectionClient(
+        getApplicationContext(), eglBase, peerConnectionParameters, CallActivity.this);
+    PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();
+    if (loopback) {
+      options.networkIgnoreMask = 0;
+    }
+    peerConnectionClient.createPeerConnectionFactory(options);
+
+    if (screencaptureEnabled) {
+      startScreenCapture();
+    } else {
+      startCall();
+    }
+  }
+
+  private DisplayMetrics getDisplayMetrics() {
+    DisplayMetrics displayMetrics = new DisplayMetrics();
+    WindowManager windowManager =
+        (WindowManager) getApplication().getSystemService(Context.WINDOW_SERVICE);
+    windowManager.getDefaultDisplay().getRealMetrics(displayMetrics);
+    return displayMetrics;
+  }
+
+  private static int getSystemUiVisibility() {
+    return View.SYSTEM_UI_FLAG_HIDE_NAVIGATION | View.SYSTEM_UI_FLAG_FULLSCREEN
+        | View.SYSTEM_UI_FLAG_IMMERSIVE_STICKY;
+  }
+
+  private void startScreenCapture() {
+    MediaProjectionManager mediaProjectionManager =
+        (MediaProjectionManager) getApplication().getSystemService(
+            Context.MEDIA_PROJECTION_SERVICE);
+    startActivityForResult(
+        mediaProjectionManager.createScreenCaptureIntent(), CAPTURE_PERMISSION_REQUEST_CODE);
+  }
+
+  @Override
+  public void onActivityResult(int requestCode, int resultCode, Intent data) {
+    if (requestCode != CAPTURE_PERMISSION_REQUEST_CODE)
+      return;
+    mediaProjectionPermissionResultCode = resultCode;
+    mediaProjectionPermissionResultData = data;
+    startCall();
+  }
+
+  private boolean useCamera2() {
+    return Camera2Enumerator.isSupported(this) && getIntent().getBooleanExtra(EXTRA_CAMERA2, true);
+  }
+
+  private boolean captureToTexture() {
+    return getIntent().getBooleanExtra(EXTRA_CAPTURETOTEXTURE_ENABLED, false);
+  }
+
+  private @Nullable VideoCapturer createCameraCapturer(CameraEnumerator enumerator) {
+    final String[] deviceNames = enumerator.getDeviceNames();
+
+    // First, try to find front facing camera
+    Logging.d(TAG, "Looking for front facing cameras.");
+    for (String deviceName : deviceNames) {
+      if (enumerator.isFrontFacing(deviceName)) {
+        Logging.d(TAG, "Creating front facing camera capturer.");
+        VideoCapturer videoCapturer = enumerator.createCapturer(deviceName, null);
+
+        if (videoCapturer != null) {
+          return videoCapturer;
+        }
+      }
+    }
+
+    // Front facing camera not found, try something else
+    Logging.d(TAG, "Looking for other cameras.");
+    for (String deviceName : deviceNames) {
+      if (!enumerator.isFrontFacing(deviceName)) {
+        Logging.d(TAG, "Creating other camera capturer.");
+        VideoCapturer videoCapturer = enumerator.createCapturer(deviceName, null);
+
+        if (videoCapturer != null) {
+          return videoCapturer;
+        }
+      }
+    }
+
+    return null;
+  }
+
+  private @Nullable VideoCapturer createScreenCapturer() {
+    if (mediaProjectionPermissionResultCode != Activity.RESULT_OK) {
+      reportError("User didn't give permission to capture the screen.");
+      return null;
+    }
+    return new ScreenCapturerAndroid(
+        mediaProjectionPermissionResultData, new MediaProjection.Callback() {
+      @Override
+      public void onStop() {
+        reportError("User revoked permission to capture the screen.");
+      }
+    });
+  }
+
+  // Activity interfaces
+  @Override
+  public void onStop() {
+    super.onStop();
+    activityRunning = false;
+    // Don't stop the video when using screencapture to allow user to show other apps to the remote
+    // end.
+    if (peerConnectionClient != null && !screencaptureEnabled) {
+      peerConnectionClient.stopVideoSource();
+    }
+    if (cpuMonitor != null) {
+      cpuMonitor.pause();
+    }
+  }
+
+  @Override
+  public void onStart() {
+    super.onStart();
+    activityRunning = true;
+    // Video is not paused for screencapture. See onPause.
+    if (peerConnectionClient != null && !screencaptureEnabled) {
+      peerConnectionClient.startVideoSource();
+    }
+    if (cpuMonitor != null) {
+      cpuMonitor.resume();
+    }
+  }
+
+  @Override
+  protected void onDestroy() {
+    Thread.setDefaultUncaughtExceptionHandler(null);
+    disconnect();
+    if (logToast != null) {
+      logToast.cancel();
+    }
+    activityRunning = false;
+    super.onDestroy();
+  }
+
+  // CallFragment.OnCallEvents interface implementation.
+  @Override
+  public void onCallHangUp() {
+    disconnect();
+  }
+
+  @Override
+  public void onCameraSwitch() {
+    if (peerConnectionClient != null) {
+      peerConnectionClient.switchCamera();
+    }
+  }
+
+  @Override
+  public void onVideoScalingSwitch(ScalingType scalingType) {
+    fullscreenRenderer.setScalingType(scalingType);
+  }
+
+  @Override
+  public void onCaptureFormatChange(int width, int height, int framerate) {
+    if (peerConnectionClient != null) {
+      peerConnectionClient.changeCaptureFormat(width, height, framerate);
+    }
+  }
+
+  @Override
+  public boolean onToggleMic() {
+    if (peerConnectionClient != null) {
+      micEnabled = !micEnabled;
+      peerConnectionClient.setAudioEnabled(micEnabled);
+    }
+    return micEnabled;
+  }
+
+  // Helper functions.
+  private void toggleCallControlFragmentVisibility() {
+    if (!connected || !callFragment.isAdded()) {
+      return;
+    }
+    // Show/hide call control fragment
+    callControlFragmentVisible = !callControlFragmentVisible;
+    FragmentTransaction ft = getFragmentManager().beginTransaction();
+    if (callControlFragmentVisible) {
+      ft.show(callFragment);
+      ft.show(hudFragment);
+    } else {
+      ft.hide(callFragment);
+      ft.hide(hudFragment);
+    }
+    ft.setTransition(FragmentTransaction.TRANSIT_FRAGMENT_FADE);
+    ft.commit();
+  }
+
+  private void startCall() {
+    if (appRtcClient == null) {
+      Log.e(TAG, "AppRTC client is not allocated for a call.");
+      return;
+    }
+    callStartedTimeMs = System.currentTimeMillis();
+
+    // Start room connection.
+    logAndToast(getString(R.string.connecting_to, roomConnectionParameters.roomUrl));
+    appRtcClient.connectToRoom(roomConnectionParameters);
+
+    // Create and audio manager that will take care of audio routing,
+    // audio modes, audio device enumeration etc.
+    audioManager = AppRTCAudioManager.create(getApplicationContext());
+    // Store existing audio settings and change audio mode to
+    // MODE_IN_COMMUNICATION for best possible VoIP performance.
+    Log.d(TAG, "Starting the audio manager...");
+    audioManager.start(new AudioManagerEvents() {
+      // This method will be called each time the number of available audio
+      // devices has changed.
+      @Override
+      public void onAudioDeviceChanged(
+          AudioDevice audioDevice, Set<AudioDevice> availableAudioDevices) {
+        onAudioManagerDevicesChanged(audioDevice, availableAudioDevices);
+      }
+    });
+  }
+
+  // Should be called from UI thread
+  private void callConnected() {
+    final long delta = System.currentTimeMillis() - callStartedTimeMs;
+    Log.i(TAG, "Call connected: delay=" + delta + "ms");
+    if (peerConnectionClient == null || isError) {
+      Log.w(TAG, "Call is connected in closed or error state");
+      return;
+    }
+    // Enable statistics callback.
+    peerConnectionClient.enableStatsEvents(true, STAT_CALLBACK_PERIOD);
+    setSwappedFeeds(false /* isSwappedFeeds */);
+  }
+
+  // This method is called when the audio manager reports audio device change,
+  // e.g. from wired headset to speakerphone.
+  private void onAudioManagerDevicesChanged(
+      final AudioDevice device, final Set<AudioDevice> availableDevices) {
+    Log.d(TAG, "onAudioManagerDevicesChanged: " + availableDevices + ", "
+            + "selected: " + device);
+    // TODO(henrika): add callback handler.
+  }
+
+  // Disconnect from remote resources, dispose of local resources, and exit.
+  private void disconnect() {
+    activityRunning = false;
+    remoteProxyRenderer.setTarget(null);
+    localProxyVideoSink.setTarget(null);
+    if (appRtcClient != null) {
+      appRtcClient.disconnectFromRoom();
+      appRtcClient = null;
+    }
+    if (pipRenderer != null) {
+      pipRenderer.release();
+      pipRenderer = null;
+    }
+    if (videoFileRenderer != null) {
+      videoFileRenderer.release();
+      videoFileRenderer = null;
+    }
+    if (fullscreenRenderer != null) {
+      fullscreenRenderer.release();
+      fullscreenRenderer = null;
+    }
+    if (peerConnectionClient != null) {
+      peerConnectionClient.close();
+      peerConnectionClient = null;
+    }
+    if (audioManager != null) {
+      audioManager.stop();
+      audioManager = null;
+    }
+    if (connected && !isError) {
+      setResult(RESULT_OK);
+    } else {
+      setResult(RESULT_CANCELED);
+    }
+    finish();
+  }
+
+  private void disconnectWithErrorMessage(final String errorMessage) {
+    if (commandLineRun || !activityRunning) {
+      Log.e(TAG, "Critical error: " + errorMessage);
+      disconnect();
+    } else {
+      new AlertDialog.Builder(this)
+          .setTitle(getText(R.string.channel_error_title))
+          .setMessage(errorMessage)
+          .setCancelable(false)
+          .setNeutralButton(R.string.ok,
+              new DialogInterface.OnClickListener() {
+                @Override
+                public void onClick(DialogInterface dialog, int id) {
+                  dialog.cancel();
+                  disconnect();
+                }
+              })
+          .create()
+          .show();
+    }
+  }
+
+  // Log `msg` and Toast about it.
+  private void logAndToast(String msg) {
+    Log.d(TAG, msg);
+    if (logToast != null) {
+      logToast.cancel();
+    }
+    logToast = Toast.makeText(this, msg, Toast.LENGTH_SHORT);
+    logToast.show();
+  }
+
+  private void reportError(final String description) {
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        if (!isError) {
+          isError = true;
+          disconnectWithErrorMessage(description);
+        }
+      }
+    });
+  }
+
+  private @Nullable VideoCapturer createVideoCapturer() {
+    final VideoCapturer videoCapturer;
+    String videoFileAsCamera = getIntent().getStringExtra(EXTRA_VIDEO_FILE_AS_CAMERA);
+    if (videoFileAsCamera != null) {
+      try {
+        videoCapturer = new FileVideoCapturer(videoFileAsCamera);
+      } catch (IOException e) {
+        reportError("Failed to open video file for emulated camera");
+        return null;
+      }
+    } else if (screencaptureEnabled) {
+      return createScreenCapturer();
+    } else if (useCamera2()) {
+      if (!captureToTexture()) {
+        reportError(getString(R.string.camera2_texture_only_error));
+        return null;
+      }
+
+      Logging.d(TAG, "Creating capturer using camera2 API.");
+      videoCapturer = createCameraCapturer(new Camera2Enumerator(this));
+    } else {
+      Logging.d(TAG, "Creating capturer using camera1 API.");
+      videoCapturer = createCameraCapturer(new Camera1Enumerator(captureToTexture()));
+    }
+    if (videoCapturer == null) {
+      reportError("Failed to open camera");
+      return null;
+    }
+    return videoCapturer;
+  }
+
+  private void setSwappedFeeds(boolean isSwappedFeeds) {
+    Logging.d(TAG, "setSwappedFeeds: " + isSwappedFeeds);
+    this.isSwappedFeeds = isSwappedFeeds;
+    localProxyVideoSink.setTarget(isSwappedFeeds ? fullscreenRenderer : pipRenderer);
+    remoteProxyRenderer.setTarget(isSwappedFeeds ? pipRenderer : fullscreenRenderer);
+    fullscreenRenderer.setMirror(isSwappedFeeds);
+    pipRenderer.setMirror(!isSwappedFeeds);
+  }
+
+  // -----Implementation of AppRTCClient.AppRTCSignalingEvents ---------------
+  // All callbacks are invoked from websocket signaling looper thread and
+  // are routed to UI thread.
+  private void onConnectedToRoomInternal(final SignalingParameters params) {
+    final long delta = System.currentTimeMillis() - callStartedTimeMs;
+
+    signalingParameters = params;
+    logAndToast("Creating peer connection, delay=" + delta + "ms");
+    VideoCapturer videoCapturer = null;
+    if (peerConnectionParameters.videoCallEnabled) {
+      videoCapturer = createVideoCapturer();
+    }
+    peerConnectionClient.createPeerConnection(
+        localProxyVideoSink, remoteSinks, videoCapturer, signalingParameters);
+
+    if (signalingParameters.initiator) {
+      logAndToast("Creating OFFER...");
+      // Create offer. Offer SDP will be sent to answering client in
+      // PeerConnectionEvents.onLocalDescription event.
+      peerConnectionClient.createOffer();
+    } else {
+      if (params.offerSdp != null) {
+        peerConnectionClient.setRemoteDescription(params.offerSdp);
+        logAndToast("Creating ANSWER...");
+        // Create answer. Answer SDP will be sent to offering client in
+        // PeerConnectionEvents.onLocalDescription event.
+        peerConnectionClient.createAnswer();
+      }
+      if (params.iceCandidates != null) {
+        // Add remote ICE candidates from room.
+        for (IceCandidate iceCandidate : params.iceCandidates) {
+          peerConnectionClient.addRemoteIceCandidate(iceCandidate);
+        }
+      }
+    }
+  }
+
+  @Override
+  public void onConnectedToRoom(final SignalingParameters params) {
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        onConnectedToRoomInternal(params);
+      }
+    });
+  }
+
+  @Override
+  public void onRemoteDescription(final SessionDescription desc) {
+    final long delta = System.currentTimeMillis() - callStartedTimeMs;
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        if (peerConnectionClient == null) {
+          Log.e(TAG, "Received remote SDP for non-initilized peer connection.");
+          return;
+        }
+        logAndToast("Received remote " + desc.type + ", delay=" + delta + "ms");
+        peerConnectionClient.setRemoteDescription(desc);
+        if (!signalingParameters.initiator) {
+          logAndToast("Creating ANSWER...");
+          // Create answer. Answer SDP will be sent to offering client in
+          // PeerConnectionEvents.onLocalDescription event.
+          peerConnectionClient.createAnswer();
+        }
+      }
+    });
+  }
+
+  @Override
+  public void onRemoteIceCandidate(final IceCandidate candidate) {
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        if (peerConnectionClient == null) {
+          Log.e(TAG, "Received ICE candidate for a non-initialized peer connection.");
+          return;
+        }
+        peerConnectionClient.addRemoteIceCandidate(candidate);
+      }
+    });
+  }
+
+  @Override
+  public void onRemoteIceCandidatesRemoved(final IceCandidate[] candidates) {
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        if (peerConnectionClient == null) {
+          Log.e(TAG, "Received ICE candidate removals for a non-initialized peer connection.");
+          return;
+        }
+        peerConnectionClient.removeRemoteIceCandidates(candidates);
+      }
+    });
+  }
+
+  @Override
+  public void onChannelClose() {
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        logAndToast("Remote end hung up; dropping PeerConnection");
+        disconnect();
+      }
+    });
+  }
+
+  @Override
+  public void onChannelError(final String description) {
+    reportError(description);
+  }
+
+  // -----Implementation of PeerConnectionClient.PeerConnectionEvents.---------
+  // Send local peer connection SDP and ICE candidates to remote party.
+  // All callbacks are invoked from peer connection client looper thread and
+  // are routed to UI thread.
+  @Override
+  public void onLocalDescription(final SessionDescription desc) {
+    final long delta = System.currentTimeMillis() - callStartedTimeMs;
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        if (appRtcClient != null) {
+          logAndToast("Sending " + desc.type + ", delay=" + delta + "ms");
+          if (signalingParameters.initiator) {
+            appRtcClient.sendOfferSdp(desc);
+          } else {
+            appRtcClient.sendAnswerSdp(desc);
+          }
+        }
+        if (peerConnectionParameters.videoMaxBitrate > 0) {
+          Log.d(TAG, "Set video maximum bitrate: " + peerConnectionParameters.videoMaxBitrate);
+          peerConnectionClient.setVideoMaxBitrate(peerConnectionParameters.videoMaxBitrate);
+        }
+      }
+    });
+  }
+
+  @Override
+  public void onIceCandidate(final IceCandidate candidate) {
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        if (appRtcClient != null) {
+          appRtcClient.sendLocalIceCandidate(candidate);
+        }
+      }
+    });
+  }
+
+  @Override
+  public void onIceCandidatesRemoved(final IceCandidate[] candidates) {
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        if (appRtcClient != null) {
+          appRtcClient.sendLocalIceCandidateRemovals(candidates);
+        }
+      }
+    });
+  }
+
+  @Override
+  public void onIceConnected() {
+    final long delta = System.currentTimeMillis() - callStartedTimeMs;
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        logAndToast("ICE connected, delay=" + delta + "ms");
+      }
+    });
+  }
+
+  @Override
+  public void onIceDisconnected() {
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        logAndToast("ICE disconnected");
+      }
+    });
+  }
+
+  @Override
+  public void onConnected() {
+    final long delta = System.currentTimeMillis() - callStartedTimeMs;
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        logAndToast("DTLS connected, delay=" + delta + "ms");
+        connected = true;
+        callConnected();
+      }
+    });
+  }
+
+  @Override
+  public void onDisconnected() {
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        logAndToast("DTLS disconnected");
+        connected = false;
+        disconnect();
+      }
+    });
+  }
+
+  @Override
+  public void onPeerConnectionClosed() {}
+
+  @Override
+  public void onPeerConnectionStatsReady(final RTCStatsReport report) {
+    runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        if (!isError && connected) {
+          hudFragment.updateEncoderStatistics(report);
+        }
+      }
+    });
+  }
+
+  @Override
+  public void onPeerConnectionError(final String description) {
+    reportError(description);
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CallFragment.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CallFragment.java
new file mode 100644
index 0000000000..0d8bdaa06f
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CallFragment.java
@@ -0,0 +1,137 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.app.Activity;
+import android.app.Fragment;
+import android.os.Bundle;
+import android.view.LayoutInflater;
+import android.view.View;
+import android.view.ViewGroup;
+import android.widget.ImageButton;
+import android.widget.SeekBar;
+import android.widget.TextView;
+
+import org.webrtc.RendererCommon.ScalingType;
+
+/**
+ * Fragment for call control.
+ */
+public class CallFragment extends Fragment {
+  private TextView contactView;
+  private ImageButton cameraSwitchButton;
+  private ImageButton videoScalingButton;
+  private ImageButton toggleMuteButton;
+  private TextView captureFormatText;
+  private SeekBar captureFormatSlider;
+  private OnCallEvents callEvents;
+  private ScalingType scalingType;
+  private boolean videoCallEnabled = true;
+
+  /**
+   * Call control interface for container activity.
+   */
+  public interface OnCallEvents {
+    void onCallHangUp();
+    void onCameraSwitch();
+    void onVideoScalingSwitch(ScalingType scalingType);
+    void onCaptureFormatChange(int width, int height, int framerate);
+    boolean onToggleMic();
+  }
+
+  @Override
+  public View onCreateView(
+      LayoutInflater inflater, ViewGroup container, Bundle savedInstanceState) {
+    View controlView = inflater.inflate(R.layout.fragment_call, container, false);
+
+    // Create UI controls.
+    contactView = controlView.findViewById(R.id.contact_name_call);
+    ImageButton disconnectButton = controlView.findViewById(R.id.button_call_disconnect);
+    cameraSwitchButton = controlView.findViewById(R.id.button_call_switch_camera);
+    videoScalingButton = controlView.findViewById(R.id.button_call_scaling_mode);
+    toggleMuteButton = controlView.findViewById(R.id.button_call_toggle_mic);
+    captureFormatText = controlView.findViewById(R.id.capture_format_text_call);
+    captureFormatSlider = controlView.findViewById(R.id.capture_format_slider_call);
+
+    // Add buttons click events.
+    disconnectButton.setOnClickListener(new View.OnClickListener() {
+      @Override
+      public void onClick(View view) {
+        callEvents.onCallHangUp();
+      }
+    });
+
+    cameraSwitchButton.setOnClickListener(new View.OnClickListener() {
+      @Override
+      public void onClick(View view) {
+        callEvents.onCameraSwitch();
+      }
+    });
+
+    videoScalingButton.setOnClickListener(new View.OnClickListener() {
+      @Override
+      public void onClick(View view) {
+        if (scalingType == ScalingType.SCALE_ASPECT_FILL) {
+          videoScalingButton.setBackgroundResource(R.drawable.ic_action_full_screen);
+          scalingType = ScalingType.SCALE_ASPECT_FIT;
+        } else {
+          videoScalingButton.setBackgroundResource(R.drawable.ic_action_return_from_full_screen);
+          scalingType = ScalingType.SCALE_ASPECT_FILL;
+        }
+        callEvents.onVideoScalingSwitch(scalingType);
+      }
+    });
+    scalingType = ScalingType.SCALE_ASPECT_FILL;
+
+    toggleMuteButton.setOnClickListener(new View.OnClickListener() {
+      @Override
+      public void onClick(View view) {
+        boolean enabled = callEvents.onToggleMic();
+        toggleMuteButton.setAlpha(enabled ? 1.0f : 0.3f);
+      }
+    });
+
+    return controlView;
+  }
+
+  @Override
+  public void onStart() {
+    super.onStart();
+
+    boolean captureSliderEnabled = false;
+    Bundle args = getArguments();
+    if (args != null) {
+      String contactName = args.getString(CallActivity.EXTRA_ROOMID);
+      contactView.setText(contactName);
+      videoCallEnabled = args.getBoolean(CallActivity.EXTRA_VIDEO_CALL, true);
+      captureSliderEnabled = videoCallEnabled
+          && args.getBoolean(CallActivity.EXTRA_VIDEO_CAPTUREQUALITYSLIDER_ENABLED, false);
+    }
+    if (!videoCallEnabled) {
+      cameraSwitchButton.setVisibility(View.INVISIBLE);
+    }
+    if (captureSliderEnabled) {
+      captureFormatSlider.setOnSeekBarChangeListener(
+          new CaptureQualityController(captureFormatText, callEvents));
+    } else {
+      captureFormatText.setVisibility(View.GONE);
+      captureFormatSlider.setVisibility(View.GONE);
+    }
+  }
+
+  // TODO(sakal): Replace with onAttach(Context) once we only support API level 23+.
+  @SuppressWarnings("deprecation")
+  @Override
+  public void onAttach(Activity activity) {
+    super.onAttach(activity);
+    callEvents = (OnCallEvents) activity;
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CaptureQualityController.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CaptureQualityController.java
new file mode 100644
index 0000000000..8a783eca9c
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CaptureQualityController.java
@@ -0,0 +1,110 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.widget.SeekBar;
+import android.widget.TextView;
+import java.util.Arrays;
+import java.util.Collections;
+import java.util.Comparator;
+import java.util.List;
+import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
+
+/**
+ * Control capture format based on a seekbar listener.
+ */
+public class CaptureQualityController implements SeekBar.OnSeekBarChangeListener {
+  private final List<CaptureFormat> formats =
+      Arrays.asList(new CaptureFormat(1280, 720, 0, 30000), new CaptureFormat(960, 540, 0, 30000),
+          new CaptureFormat(640, 480, 0, 30000), new CaptureFormat(480, 360, 0, 30000),
+          new CaptureFormat(320, 240, 0, 30000), new CaptureFormat(256, 144, 0, 30000));
+  // Prioritize framerate below this threshold and resolution above the threshold.
+  private static final int FRAMERATE_THRESHOLD = 15;
+  private TextView captureFormatText;
+  private CallFragment.OnCallEvents callEvents;
+  private int width;
+  private int height;
+  private int framerate;
+  private double targetBandwidth;
+
+  public CaptureQualityController(
+      TextView captureFormatText, CallFragment.OnCallEvents callEvents) {
+    this.captureFormatText = captureFormatText;
+    this.callEvents = callEvents;
+  }
+
+  private final Comparator<CaptureFormat> compareFormats = new Comparator<CaptureFormat>() {
+    @Override
+    public int compare(CaptureFormat first, CaptureFormat second) {
+      int firstFps = calculateFramerate(targetBandwidth, first);
+      int secondFps = calculateFramerate(targetBandwidth, second);
+
+      if ((firstFps >= FRAMERATE_THRESHOLD && secondFps >= FRAMERATE_THRESHOLD)
+          || firstFps == secondFps) {
+        // Compare resolution.
+        return first.width * first.height - second.width * second.height;
+      } else {
+        // Compare fps.
+        return firstFps - secondFps;
+      }
+    }
+  };
+
+  @Override
+  public void onProgressChanged(SeekBar seekBar, int progress, boolean fromUser) {
+    if (progress == 0) {
+      width = 0;
+      height = 0;
+      framerate = 0;
+      captureFormatText.setText(R.string.muted);
+      return;
+    }
+
+    // Extract max bandwidth (in millipixels / second).
+    long maxCaptureBandwidth = java.lang.Long.MIN_VALUE;
+    for (CaptureFormat format : formats) {
+      maxCaptureBandwidth =
+          Math.max(maxCaptureBandwidth, (long) format.width * format.height * format.framerate.max);
+    }
+
+    // Fraction between 0 and 1.
+    double bandwidthFraction = (double) progress / 100.0;
+    // Make a log-scale transformation, still between 0 and 1.
+    final double kExpConstant = 3.0;
+    bandwidthFraction =
+        (Math.exp(kExpConstant * bandwidthFraction) - 1) / (Math.exp(kExpConstant) - 1);
+    targetBandwidth = bandwidthFraction * maxCaptureBandwidth;
+
+    // Choose the best format given a target bandwidth.
+    final CaptureFormat bestFormat = Collections.max(formats, compareFormats);
+    width = bestFormat.width;
+    height = bestFormat.height;
+    framerate = calculateFramerate(targetBandwidth, bestFormat);
+    captureFormatText.setText(
+        String.format(captureFormatText.getContext().getString(R.string.format_description), width,
+            height, framerate));
+  }
+
+  @Override
+  public void onStartTrackingTouch(SeekBar seekBar) {}
+
+  @Override
+  public void onStopTrackingTouch(SeekBar seekBar) {
+    callEvents.onCaptureFormatChange(width, height, framerate);
+  }
+
+  // Return the highest frame rate possible based on bandwidth and format.
+  private int calculateFramerate(double bandwidth, CaptureFormat format) {
+    return (int) Math.round(
+        Math.min(format.framerate.max, (int) Math.round(bandwidth / (format.width * format.height)))
+        / 1000.0);
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/ConnectActivity.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/ConnectActivity.java
new file mode 100644
index 0000000000..7206c88498
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/ConnectActivity.java
@@ -0,0 +1,666 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.annotation.TargetApi;
+import android.app.Activity;
+import android.app.AlertDialog;
+import android.content.DialogInterface;
+import android.content.Intent;
+import android.content.SharedPreferences;
+import android.content.pm.PackageInfo;
+import android.content.pm.PackageManager;
+import android.net.Uri;
+import android.os.Build;
+import android.os.Bundle;
+import android.preference.PreferenceManager;
+import android.util.Log;
+import android.view.ContextMenu;
+import android.view.KeyEvent;
+import android.view.Menu;
+import android.view.MenuItem;
+import android.view.View;
+import android.view.View.OnClickListener;
+import android.view.inputmethod.EditorInfo;
+import android.webkit.URLUtil;
+import android.widget.AdapterView;
+import android.widget.ArrayAdapter;
+import android.widget.EditText;
+import android.widget.ImageButton;
+import android.widget.ListView;
+import android.widget.TextView;
+import androidx.annotation.Nullable;
+import java.util.ArrayList;
+import java.util.Random;
+import org.json.JSONArray;
+import org.json.JSONException;
+
+/**
+ * Handles the initial setup where the user selects which room to join.
+ */
+public class ConnectActivity extends Activity {
+  private static final String TAG = "ConnectActivity";
+  private static final int CONNECTION_REQUEST = 1;
+  private static final int PERMISSION_REQUEST = 2;
+  private static final int REMOVE_FAVORITE_INDEX = 0;
+  private static boolean commandLineRun;
+
+  private ImageButton addFavoriteButton;
+  private EditText roomEditText;
+  private ListView roomListView;
+  private SharedPreferences sharedPref;
+  private String keyprefResolution;
+  private String keyprefFps;
+  private String keyprefVideoBitrateType;
+  private String keyprefVideoBitrateValue;
+  private String keyprefAudioBitrateType;
+  private String keyprefAudioBitrateValue;
+  private String keyprefRoomServerUrl;
+  private String keyprefRoom;
+  private String keyprefRoomList;
+  private ArrayList<String> roomList;
+  private ArrayAdapter<String> adapter;
+
+  @Override
+  public void onCreate(Bundle savedInstanceState) {
+    super.onCreate(savedInstanceState);
+
+    // Get setting keys.
+    PreferenceManager.setDefaultValues(this, R.xml.preferences, false);
+    sharedPref = PreferenceManager.getDefaultSharedPreferences(this);
+    keyprefResolution = getString(R.string.pref_resolution_key);
+    keyprefFps = getString(R.string.pref_fps_key);
+    keyprefVideoBitrateType = getString(R.string.pref_maxvideobitrate_key);
+    keyprefVideoBitrateValue = getString(R.string.pref_maxvideobitratevalue_key);
+    keyprefAudioBitrateType = getString(R.string.pref_startaudiobitrate_key);
+    keyprefAudioBitrateValue = getString(R.string.pref_startaudiobitratevalue_key);
+    keyprefRoomServerUrl = getString(R.string.pref_room_server_url_key);
+    keyprefRoom = getString(R.string.pref_room_key);
+    keyprefRoomList = getString(R.string.pref_room_list_key);
+
+    setContentView(R.layout.activity_connect);
+
+    roomEditText = findViewById(R.id.room_edittext);
+    roomEditText.setOnEditorActionListener(new TextView.OnEditorActionListener() {
+      @Override
+      public boolean onEditorAction(TextView textView, int i, KeyEvent keyEvent) {
+        if (i == EditorInfo.IME_ACTION_DONE) {
+          addFavoriteButton.performClick();
+          return true;
+        }
+        return false;
+      }
+    });
+    roomEditText.requestFocus();
+
+    roomListView = findViewById(R.id.room_listview);
+    roomListView.setEmptyView(findViewById(android.R.id.empty));
+    roomListView.setOnItemClickListener(roomListClickListener);
+    registerForContextMenu(roomListView);
+    ImageButton connectButton = findViewById(R.id.connect_button);
+    connectButton.setOnClickListener(connectListener);
+    addFavoriteButton = findViewById(R.id.add_favorite_button);
+    addFavoriteButton.setOnClickListener(addFavoriteListener);
+
+    requestPermissions();
+  }
+
+  @Override
+  public boolean onCreateOptionsMenu(Menu menu) {
+    getMenuInflater().inflate(R.menu.connect_menu, menu);
+    return true;
+  }
+
+  @Override
+  public void onCreateContextMenu(ContextMenu menu, View v, ContextMenu.ContextMenuInfo menuInfo) {
+    if (v.getId() == R.id.room_listview) {
+      AdapterView.AdapterContextMenuInfo info = (AdapterView.AdapterContextMenuInfo) menuInfo;
+      menu.setHeaderTitle(roomList.get(info.position));
+      String[] menuItems = getResources().getStringArray(R.array.roomListContextMenu);
+      for (int i = 0; i < menuItems.length; i++) {
+        menu.add(Menu.NONE, i, i, menuItems[i]);
+      }
+    } else {
+      super.onCreateContextMenu(menu, v, menuInfo);
+    }
+  }
+
+  @Override
+  public boolean onContextItemSelected(MenuItem item) {
+    if (item.getItemId() == REMOVE_FAVORITE_INDEX) {
+      AdapterView.AdapterContextMenuInfo info =
+          (AdapterView.AdapterContextMenuInfo) item.getMenuInfo();
+      roomList.remove(info.position);
+      adapter.notifyDataSetChanged();
+      return true;
+    }
+
+    return super.onContextItemSelected(item);
+  }
+
+  @Override
+  public boolean onOptionsItemSelected(MenuItem item) {
+    // Handle presses on the action bar items.
+    if (item.getItemId() == R.id.action_settings) {
+      Intent intent = new Intent(this, SettingsActivity.class);
+      startActivity(intent);
+      return true;
+    } else if (item.getItemId() == R.id.action_loopback) {
+      connectToRoom(null, false, true, false, 0);
+      return true;
+    } else {
+      return super.onOptionsItemSelected(item);
+    }
+  }
+
+  @Override
+  public void onPause() {
+    super.onPause();
+    String room = roomEditText.getText().toString();
+    String roomListJson = new JSONArray(roomList).toString();
+    SharedPreferences.Editor editor = sharedPref.edit();
+    editor.putString(keyprefRoom, room);
+    editor.putString(keyprefRoomList, roomListJson);
+    editor.commit();
+  }
+
+  @Override
+  public void onResume() {
+    super.onResume();
+    String room = sharedPref.getString(keyprefRoom, "");
+    roomEditText.setText(room);
+    roomList = new ArrayList<>();
+    String roomListJson = sharedPref.getString(keyprefRoomList, null);
+    if (roomListJson != null) {
+      try {
+        JSONArray jsonArray = new JSONArray(roomListJson);
+        for (int i = 0; i < jsonArray.length(); i++) {
+          roomList.add(jsonArray.get(i).toString());
+        }
+      } catch (JSONException e) {
+        Log.e(TAG, "Failed to load room list: " + e.toString());
+      }
+    }
+    adapter = new ArrayAdapter<>(this, android.R.layout.simple_list_item_1, roomList);
+    roomListView.setAdapter(adapter);
+    if (adapter.getCount() > 0) {
+      roomListView.requestFocus();
+      roomListView.setItemChecked(0, true);
+    }
+  }
+
+  @Override
+  protected void onActivityResult(int requestCode, int resultCode, Intent data) {
+    if (requestCode == CONNECTION_REQUEST && commandLineRun) {
+      Log.d(TAG, "Return: " + resultCode);
+      setResult(resultCode);
+      commandLineRun = false;
+      finish();
+    }
+  }
+
+  @Override
+  public void onRequestPermissionsResult(
+      int requestCode, String[] permissions, int[] grantResults) {
+    if (requestCode == PERMISSION_REQUEST) {
+      String[] missingPermissions = getMissingPermissions();
+      if (missingPermissions.length != 0) {
+        // User didn't grant all the permissions. Warn that the application might not work
+        // correctly.
+        new AlertDialog.Builder(this)
+            .setMessage(R.string.missing_permissions_try_again)
+            .setPositiveButton(R.string.yes,
+                (dialog, id) -> {
+                  // User wants to try giving the permissions again.
+                  dialog.cancel();
+                  requestPermissions();
+                })
+            .setNegativeButton(R.string.no,
+                (dialog, id) -> {
+                  // User doesn't want to give the permissions.
+                  dialog.cancel();
+                  onPermissionsGranted();
+                })
+            .show();
+      } else {
+        // All permissions granted.
+        onPermissionsGranted();
+      }
+    }
+  }
+
+  private void onPermissionsGranted() {
+    // If an implicit VIEW intent is launching the app, go directly to that URL.
+    final Intent intent = getIntent();
+    if ("android.intent.action.VIEW".equals(intent.getAction()) && !commandLineRun) {
+      boolean loopback = intent.getBooleanExtra(CallActivity.EXTRA_LOOPBACK, false);
+      int runTimeMs = intent.getIntExtra(CallActivity.EXTRA_RUNTIME, 0);
+      boolean useValuesFromIntent =
+          intent.getBooleanExtra(CallActivity.EXTRA_USE_VALUES_FROM_INTENT, false);
+      String room = sharedPref.getString(keyprefRoom, "");
+      connectToRoom(room, true, loopback, useValuesFromIntent, runTimeMs);
+    }
+  }
+
+  @TargetApi(Build.VERSION_CODES.M)
+  private void requestPermissions() {
+    if (Build.VERSION.SDK_INT < Build.VERSION_CODES.M) {
+      // Dynamic permissions are not required before Android M.
+      onPermissionsGranted();
+      return;
+    }
+
+    String[] missingPermissions = getMissingPermissions();
+    if (missingPermissions.length != 0) {
+      requestPermissions(missingPermissions, PERMISSION_REQUEST);
+    } else {
+      onPermissionsGranted();
+    }
+  }
+
+  @TargetApi(Build.VERSION_CODES.M)
+  private String[] getMissingPermissions() {
+    if (Build.VERSION.SDK_INT < Build.VERSION_CODES.M) {
+      return new String[0];
+    }
+
+    PackageInfo info;
+    try {
+      info = getPackageManager().getPackageInfo(getPackageName(), PackageManager.GET_PERMISSIONS);
+    } catch (PackageManager.NameNotFoundException e) {
+      Log.w(TAG, "Failed to retrieve permissions.");
+      return new String[0];
+    }
+
+    if (info.requestedPermissions == null) {
+      Log.w(TAG, "No requested permissions.");
+      return new String[0];
+    }
+
+    ArrayList<String> missingPermissions = new ArrayList<>();
+    for (int i = 0; i < info.requestedPermissions.length; i++) {
+      if ((info.requestedPermissionsFlags[i] & PackageInfo.REQUESTED_PERMISSION_GRANTED) == 0) {
+        missingPermissions.add(info.requestedPermissions[i]);
+      }
+    }
+    Log.d(TAG, "Missing permissions: " + missingPermissions);
+
+    return missingPermissions.toArray(new String[missingPermissions.size()]);
+  }
+
+  /**
+   * Get a value from the shared preference or from the intent, if it does not
+   * exist the default is used.
+   */
+  @Nullable
+  private String sharedPrefGetString(
+      int attributeId, String intentName, int defaultId, boolean useFromIntent) {
+    String defaultValue = getString(defaultId);
+    if (useFromIntent) {
+      String value = getIntent().getStringExtra(intentName);
+      if (value != null) {
+        return value;
+      }
+      return defaultValue;
+    } else {
+      String attributeName = getString(attributeId);
+      return sharedPref.getString(attributeName, defaultValue);
+    }
+  }
+
+  /**
+   * Get a value from the shared preference or from the intent, if it does not
+   * exist the default is used.
+   */
+  private boolean sharedPrefGetBoolean(
+      int attributeId, String intentName, int defaultId, boolean useFromIntent) {
+    boolean defaultValue = Boolean.parseBoolean(getString(defaultId));
+    if (useFromIntent) {
+      return getIntent().getBooleanExtra(intentName, defaultValue);
+    } else {
+      String attributeName = getString(attributeId);
+      return sharedPref.getBoolean(attributeName, defaultValue);
+    }
+  }
+
+  /**
+   * Get a value from the shared preference or from the intent, if it does not
+   * exist the default is used.
+   */
+  private int sharedPrefGetInteger(
+      int attributeId, String intentName, int defaultId, boolean useFromIntent) {
+    String defaultString = getString(defaultId);
+    int defaultValue = Integer.parseInt(defaultString);
+    if (useFromIntent) {
+      return getIntent().getIntExtra(intentName, defaultValue);
+    } else {
+      String attributeName = getString(attributeId);
+      String value = sharedPref.getString(attributeName, defaultString);
+      try {
+        return Integer.parseInt(value);
+      } catch (NumberFormatException e) {
+        Log.e(TAG, "Wrong setting for: " + attributeName + ":" + value);
+        return defaultValue;
+      }
+    }
+  }
+
+  @SuppressWarnings("StringSplitter")
+  private void connectToRoom(String roomId, boolean commandLineRun, boolean loopback,
+      boolean useValuesFromIntent, int runTimeMs) {
+    ConnectActivity.commandLineRun = commandLineRun;
+
+    // roomId is random for loopback.
+    if (loopback) {
+      roomId = Integer.toString((new Random()).nextInt(100000000));
+    }
+
+    String roomUrl = sharedPref.getString(
+        keyprefRoomServerUrl, getString(R.string.pref_room_server_url_default));
+
+    // Video call enabled flag.
+    boolean videoCallEnabled = sharedPrefGetBoolean(R.string.pref_videocall_key,
+        CallActivity.EXTRA_VIDEO_CALL, R.string.pref_videocall_default, useValuesFromIntent);
+
+    // Use screencapture option.
+    boolean useScreencapture = sharedPrefGetBoolean(R.string.pref_screencapture_key,
+        CallActivity.EXTRA_SCREENCAPTURE, R.string.pref_screencapture_default, useValuesFromIntent);
+
+    // Use Camera2 option.
+    boolean useCamera2 = sharedPrefGetBoolean(R.string.pref_camera2_key, CallActivity.EXTRA_CAMERA2,
+        R.string.pref_camera2_default, useValuesFromIntent);
+
+    // Get default codecs.
+    String videoCodec = sharedPrefGetString(R.string.pref_videocodec_key,
+        CallActivity.EXTRA_VIDEOCODEC, R.string.pref_videocodec_default, useValuesFromIntent);
+    String audioCodec = sharedPrefGetString(R.string.pref_audiocodec_key,
+        CallActivity.EXTRA_AUDIOCODEC, R.string.pref_audiocodec_default, useValuesFromIntent);
+
+    // Check HW codec flag.
+    boolean hwCodec = sharedPrefGetBoolean(R.string.pref_hwcodec_key,
+        CallActivity.EXTRA_HWCODEC_ENABLED, R.string.pref_hwcodec_default, useValuesFromIntent);
+
+    // Check Capture to texture.
+    boolean captureToTexture = sharedPrefGetBoolean(R.string.pref_capturetotexture_key,
+        CallActivity.EXTRA_CAPTURETOTEXTURE_ENABLED, R.string.pref_capturetotexture_default,
+        useValuesFromIntent);
+
+    // Check FlexFEC.
+    boolean flexfecEnabled = sharedPrefGetBoolean(R.string.pref_flexfec_key,
+        CallActivity.EXTRA_FLEXFEC_ENABLED, R.string.pref_flexfec_default, useValuesFromIntent);
+
+    // Check Disable Audio Processing flag.
+    boolean noAudioProcessing = sharedPrefGetBoolean(R.string.pref_noaudioprocessing_key,
+        CallActivity.EXTRA_NOAUDIOPROCESSING_ENABLED, R.string.pref_noaudioprocessing_default,
+        useValuesFromIntent);
+
+    boolean aecDump = sharedPrefGetBoolean(R.string.pref_aecdump_key,
+        CallActivity.EXTRA_AECDUMP_ENABLED, R.string.pref_aecdump_default, useValuesFromIntent);
+
+    boolean saveInputAudioToFile =
+        sharedPrefGetBoolean(R.string.pref_enable_save_input_audio_to_file_key,
+            CallActivity.EXTRA_SAVE_INPUT_AUDIO_TO_FILE_ENABLED,
+            R.string.pref_enable_save_input_audio_to_file_default, useValuesFromIntent);
+
+    // Check OpenSL ES enabled flag.
+    boolean useOpenSLES = sharedPrefGetBoolean(R.string.pref_opensles_key,
+        CallActivity.EXTRA_OPENSLES_ENABLED, R.string.pref_opensles_default, useValuesFromIntent);
+
+    // Check Disable built-in AEC flag.
+    boolean disableBuiltInAEC = sharedPrefGetBoolean(R.string.pref_disable_built_in_aec_key,
+        CallActivity.EXTRA_DISABLE_BUILT_IN_AEC, R.string.pref_disable_built_in_aec_default,
+        useValuesFromIntent);
+
+    // Check Disable built-in AGC flag.
+    boolean disableBuiltInAGC = sharedPrefGetBoolean(R.string.pref_disable_built_in_agc_key,
+        CallActivity.EXTRA_DISABLE_BUILT_IN_AGC, R.string.pref_disable_built_in_agc_default,
+        useValuesFromIntent);
+
+    // Check Disable built-in NS flag.
+    boolean disableBuiltInNS = sharedPrefGetBoolean(R.string.pref_disable_built_in_ns_key,
+        CallActivity.EXTRA_DISABLE_BUILT_IN_NS, R.string.pref_disable_built_in_ns_default,
+        useValuesFromIntent);
+
+    // Check Disable gain control
+    boolean disableWebRtcAGCAndHPF = sharedPrefGetBoolean(
+        R.string.pref_disable_webrtc_agc_and_hpf_key, CallActivity.EXTRA_DISABLE_WEBRTC_AGC_AND_HPF,
+        R.string.pref_disable_webrtc_agc_and_hpf_key, useValuesFromIntent);
+
+    // Get video resolution from settings.
+    int videoWidth = 0;
+    int videoHeight = 0;
+    if (useValuesFromIntent) {
+      videoWidth = getIntent().getIntExtra(CallActivity.EXTRA_VIDEO_WIDTH, 0);
+      videoHeight = getIntent().getIntExtra(CallActivity.EXTRA_VIDEO_HEIGHT, 0);
+    }
+    if (videoWidth == 0 && videoHeight == 0) {
+      String resolution =
+          sharedPref.getString(keyprefResolution, getString(R.string.pref_resolution_default));
+      String[] dimensions = resolution.split("[ x]+");
+      if (dimensions.length == 2) {
+        try {
+          videoWidth = Integer.parseInt(dimensions[0]);
+          videoHeight = Integer.parseInt(dimensions[1]);
+        } catch (NumberFormatException e) {
+          videoWidth = 0;
+          videoHeight = 0;
+          Log.e(TAG, "Wrong video resolution setting: " + resolution);
+        }
+      }
+    }
+
+    // Get camera fps from settings.
+    int cameraFps = 0;
+    if (useValuesFromIntent) {
+      cameraFps = getIntent().getIntExtra(CallActivity.EXTRA_VIDEO_FPS, 0);
+    }
+    if (cameraFps == 0) {
+      String fps = sharedPref.getString(keyprefFps, getString(R.string.pref_fps_default));
+      String[] fpsValues = fps.split("[ x]+");
+      if (fpsValues.length == 2) {
+        try {
+          cameraFps = Integer.parseInt(fpsValues[0]);
+        } catch (NumberFormatException e) {
+          cameraFps = 0;
+          Log.e(TAG, "Wrong camera fps setting: " + fps);
+        }
+      }
+    }
+
+    // Check capture quality slider flag.
+    boolean captureQualitySlider = sharedPrefGetBoolean(R.string.pref_capturequalityslider_key,
+        CallActivity.EXTRA_VIDEO_CAPTUREQUALITYSLIDER_ENABLED,
+        R.string.pref_capturequalityslider_default, useValuesFromIntent);
+
+    // Get video and audio start bitrate.
+    int videoStartBitrate = 0;
+    if (useValuesFromIntent) {
+      videoStartBitrate = getIntent().getIntExtra(CallActivity.EXTRA_VIDEO_BITRATE, 0);
+    }
+    if (videoStartBitrate == 0) {
+      String bitrateTypeDefault = getString(R.string.pref_maxvideobitrate_default);
+      String bitrateType = sharedPref.getString(keyprefVideoBitrateType, bitrateTypeDefault);
+      if (!bitrateType.equals(bitrateTypeDefault)) {
+        String bitrateValue = sharedPref.getString(
+            keyprefVideoBitrateValue, getString(R.string.pref_maxvideobitratevalue_default));
+        videoStartBitrate = Integer.parseInt(bitrateValue);
+      }
+    }
+
+    int audioStartBitrate = 0;
+    if (useValuesFromIntent) {
+      audioStartBitrate = getIntent().getIntExtra(CallActivity.EXTRA_AUDIO_BITRATE, 0);
+    }
+    if (audioStartBitrate == 0) {
+      String bitrateTypeDefault = getString(R.string.pref_startaudiobitrate_default);
+      String bitrateType = sharedPref.getString(keyprefAudioBitrateType, bitrateTypeDefault);
+      if (!bitrateType.equals(bitrateTypeDefault)) {
+        String bitrateValue = sharedPref.getString(
+            keyprefAudioBitrateValue, getString(R.string.pref_startaudiobitratevalue_default));
+        audioStartBitrate = Integer.parseInt(bitrateValue);
+      }
+    }
+
+    // Check statistics display option.
+    boolean displayHud = sharedPrefGetBoolean(R.string.pref_displayhud_key,
+        CallActivity.EXTRA_DISPLAY_HUD, R.string.pref_displayhud_default, useValuesFromIntent);
+
+    boolean tracing = sharedPrefGetBoolean(R.string.pref_tracing_key, CallActivity.EXTRA_TRACING,
+        R.string.pref_tracing_default, useValuesFromIntent);
+
+    // Check Enable RtcEventLog.
+    boolean rtcEventLogEnabled = sharedPrefGetBoolean(R.string.pref_enable_rtceventlog_key,
+        CallActivity.EXTRA_ENABLE_RTCEVENTLOG, R.string.pref_enable_rtceventlog_default,
+        useValuesFromIntent);
+
+    // Get datachannel options
+    boolean dataChannelEnabled = sharedPrefGetBoolean(R.string.pref_enable_datachannel_key,
+        CallActivity.EXTRA_DATA_CHANNEL_ENABLED, R.string.pref_enable_datachannel_default,
+        useValuesFromIntent);
+    boolean ordered = sharedPrefGetBoolean(R.string.pref_ordered_key, CallActivity.EXTRA_ORDERED,
+        R.string.pref_ordered_default, useValuesFromIntent);
+    boolean negotiated = sharedPrefGetBoolean(R.string.pref_negotiated_key,
+        CallActivity.EXTRA_NEGOTIATED, R.string.pref_negotiated_default, useValuesFromIntent);
+    int maxRetrMs = sharedPrefGetInteger(R.string.pref_max_retransmit_time_ms_key,
+        CallActivity.EXTRA_MAX_RETRANSMITS_MS, R.string.pref_max_retransmit_time_ms_default,
+        useValuesFromIntent);
+    int maxRetr =
+        sharedPrefGetInteger(R.string.pref_max_retransmits_key, CallActivity.EXTRA_MAX_RETRANSMITS,
+            R.string.pref_max_retransmits_default, useValuesFromIntent);
+    int id = sharedPrefGetInteger(R.string.pref_data_id_key, CallActivity.EXTRA_ID,
+        R.string.pref_data_id_default, useValuesFromIntent);
+    String protocol = sharedPrefGetString(R.string.pref_data_protocol_key,
+        CallActivity.EXTRA_PROTOCOL, R.string.pref_data_protocol_default, useValuesFromIntent);
+
+    // Start AppRTCMobile activity.
+    Log.d(TAG, "Connecting to room " + roomId + " at URL " + roomUrl);
+    if (validateUrl(roomUrl)) {
+      Uri uri = Uri.parse(roomUrl);
+      Intent intent = new Intent(this, CallActivity.class);
+      intent.setData(uri);
+      intent.putExtra(CallActivity.EXTRA_ROOMID, roomId);
+      intent.putExtra(CallActivity.EXTRA_LOOPBACK, loopback);
+      intent.putExtra(CallActivity.EXTRA_VIDEO_CALL, videoCallEnabled);
+      intent.putExtra(CallActivity.EXTRA_SCREENCAPTURE, useScreencapture);
+      intent.putExtra(CallActivity.EXTRA_CAMERA2, useCamera2);
+      intent.putExtra(CallActivity.EXTRA_VIDEO_WIDTH, videoWidth);
+      intent.putExtra(CallActivity.EXTRA_VIDEO_HEIGHT, videoHeight);
+      intent.putExtra(CallActivity.EXTRA_VIDEO_FPS, cameraFps);
+      intent.putExtra(CallActivity.EXTRA_VIDEO_CAPTUREQUALITYSLIDER_ENABLED, captureQualitySlider);
+      intent.putExtra(CallActivity.EXTRA_VIDEO_BITRATE, videoStartBitrate);
+      intent.putExtra(CallActivity.EXTRA_VIDEOCODEC, videoCodec);
+      intent.putExtra(CallActivity.EXTRA_HWCODEC_ENABLED, hwCodec);
+      intent.putExtra(CallActivity.EXTRA_CAPTURETOTEXTURE_ENABLED, captureToTexture);
+      intent.putExtra(CallActivity.EXTRA_FLEXFEC_ENABLED, flexfecEnabled);
+      intent.putExtra(CallActivity.EXTRA_NOAUDIOPROCESSING_ENABLED, noAudioProcessing);
+      intent.putExtra(CallActivity.EXTRA_AECDUMP_ENABLED, aecDump);
+      intent.putExtra(CallActivity.EXTRA_SAVE_INPUT_AUDIO_TO_FILE_ENABLED, saveInputAudioToFile);
+      intent.putExtra(CallActivity.EXTRA_OPENSLES_ENABLED, useOpenSLES);
+      intent.putExtra(CallActivity.EXTRA_DISABLE_BUILT_IN_AEC, disableBuiltInAEC);
+      intent.putExtra(CallActivity.EXTRA_DISABLE_BUILT_IN_AGC, disableBuiltInAGC);
+      intent.putExtra(CallActivity.EXTRA_DISABLE_BUILT_IN_NS, disableBuiltInNS);
+      intent.putExtra(CallActivity.EXTRA_DISABLE_WEBRTC_AGC_AND_HPF, disableWebRtcAGCAndHPF);
+      intent.putExtra(CallActivity.EXTRA_AUDIO_BITRATE, audioStartBitrate);
+      intent.putExtra(CallActivity.EXTRA_AUDIOCODEC, audioCodec);
+      intent.putExtra(CallActivity.EXTRA_DISPLAY_HUD, displayHud);
+      intent.putExtra(CallActivity.EXTRA_TRACING, tracing);
+      intent.putExtra(CallActivity.EXTRA_ENABLE_RTCEVENTLOG, rtcEventLogEnabled);
+      intent.putExtra(CallActivity.EXTRA_CMDLINE, commandLineRun);
+      intent.putExtra(CallActivity.EXTRA_RUNTIME, runTimeMs);
+      intent.putExtra(CallActivity.EXTRA_DATA_CHANNEL_ENABLED, dataChannelEnabled);
+
+      if (dataChannelEnabled) {
+        intent.putExtra(CallActivity.EXTRA_ORDERED, ordered);
+        intent.putExtra(CallActivity.EXTRA_MAX_RETRANSMITS_MS, maxRetrMs);
+        intent.putExtra(CallActivity.EXTRA_MAX_RETRANSMITS, maxRetr);
+        intent.putExtra(CallActivity.EXTRA_PROTOCOL, protocol);
+        intent.putExtra(CallActivity.EXTRA_NEGOTIATED, negotiated);
+        intent.putExtra(CallActivity.EXTRA_ID, id);
+      }
+
+      if (useValuesFromIntent) {
+        if (getIntent().hasExtra(CallActivity.EXTRA_VIDEO_FILE_AS_CAMERA)) {
+          String videoFileAsCamera =
+              getIntent().getStringExtra(CallActivity.EXTRA_VIDEO_FILE_AS_CAMERA);
+          intent.putExtra(CallActivity.EXTRA_VIDEO_FILE_AS_CAMERA, videoFileAsCamera);
+        }
+
+        if (getIntent().hasExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE)) {
+          String saveRemoteVideoToFile =
+              getIntent().getStringExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE);
+          intent.putExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE, saveRemoteVideoToFile);
+        }
+
+        if (getIntent().hasExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_WIDTH)) {
+          int videoOutWidth =
+              getIntent().getIntExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_WIDTH, 0);
+          intent.putExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_WIDTH, videoOutWidth);
+        }
+
+        if (getIntent().hasExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT)) {
+          int videoOutHeight =
+              getIntent().getIntExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT, 0);
+          intent.putExtra(CallActivity.EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT, videoOutHeight);
+        }
+      }
+
+      startActivityForResult(intent, CONNECTION_REQUEST);
+    }
+  }
+
+  private boolean validateUrl(String url) {
+    if (URLUtil.isHttpsUrl(url) || URLUtil.isHttpUrl(url)) {
+      return true;
+    }
+
+    new AlertDialog.Builder(this)
+        .setTitle(getText(R.string.invalid_url_title))
+        .setMessage(getString(R.string.invalid_url_text, url))
+        .setCancelable(false)
+        .setNeutralButton(R.string.ok,
+            new DialogInterface.OnClickListener() {
+              @Override
+              public void onClick(DialogInterface dialog, int id) {
+                dialog.cancel();
+              }
+            })
+        .create()
+        .show();
+    return false;
+  }
+
+  private final AdapterView.OnItemClickListener roomListClickListener =
+      new AdapterView.OnItemClickListener() {
+        @Override
+        public void onItemClick(AdapterView<?> adapterView, View view, int i, long l) {
+          String roomId = ((TextView) view).getText().toString();
+          connectToRoom(roomId, false, false, false, 0);
+        }
+      };
+
+  private final OnClickListener addFavoriteListener = new OnClickListener() {
+    @Override
+    public void onClick(View view) {
+      String newRoom = roomEditText.getText().toString();
+      if (newRoom.length() > 0 && !roomList.contains(newRoom)) {
+        adapter.add(newRoom);
+        adapter.notifyDataSetChanged();
+      }
+    }
+  };
+
+  private final OnClickListener connectListener = new OnClickListener() {
+    @Override
+    public void onClick(View view) {
+      connectToRoom(roomEditText.getText().toString(), false, false, false, 0);
+    }
+  };
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CpuMonitor.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CpuMonitor.java
new file mode 100644
index 0000000000..1c64621864
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/CpuMonitor.java
@@ -0,0 +1,521 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.annotation.TargetApi;
+import android.content.Context;
+import android.content.Intent;
+import android.content.IntentFilter;
+import android.os.BatteryManager;
+import android.os.Build;
+import android.os.SystemClock;
+import android.util.Log;
+import androidx.annotation.Nullable;
+import java.io.BufferedReader;
+import java.io.FileInputStream;
+import java.io.FileNotFoundException;
+import java.io.IOException;
+import java.io.InputStreamReader;
+import java.nio.charset.Charset;
+import java.util.Arrays;
+import java.util.Scanner;
+import java.util.concurrent.Executors;
+import java.util.concurrent.Future;
+import java.util.concurrent.ScheduledExecutorService;
+import java.util.concurrent.TimeUnit;
+
+/**
+ * Simple CPU monitor.  The caller creates a CpuMonitor object which can then
+ * be used via sampleCpuUtilization() to collect the percentual use of the
+ * cumulative CPU capacity for all CPUs running at their nominal frequency.  3
+ * values are generated: (1) getCpuCurrent() returns the use since the last
+ * sampleCpuUtilization(), (2) getCpuAvg3() returns the use since 3 prior
+ * calls, and (3) getCpuAvgAll() returns the use over all SAMPLE_SAVE_NUMBER
+ * calls.
+ *
+ * <p>CPUs in Android are often "offline", and while this of course means 0 Hz
+ * as current frequency, in this state we cannot even get their nominal
+ * frequency.  We therefore tread carefully, and allow any CPU to be missing.
+ * Missing CPUs are assumed to have the same nominal frequency as any close
+ * lower-numbered CPU, but as soon as it is online, we'll get their proper
+ * frequency and remember it.  (Since CPU 0 in practice always seem to be
+ * online, this unidirectional frequency inheritance should be no problem in
+ * practice.)
+ *
+ * <p>Caveats:
+ *   o No provision made for zany "turbo" mode, common in the x86 world.
+ *   o No provision made for ARM big.LITTLE; if CPU n can switch behind our
+ *     back, we might get incorrect estimates.
+ *   o This is not thread-safe.  To call asynchronously, create different
+ *     CpuMonitor objects.
+ *
+ * <p>If we can gather enough info to generate a sensible result,
+ * sampleCpuUtilization returns true.  It is designed to never throw an
+ * exception.
+ *
+ * <p>sampleCpuUtilization should not be called too often in its present form,
+ * since then deltas would be small and the percent values would fluctuate and
+ * be unreadable. If it is desirable to call it more often than say once per
+ * second, one would need to increase SAMPLE_SAVE_NUMBER and probably use
+ * Queue<Integer> to avoid copying overhead.
+ *
+ * <p>Known problems:
+ *   1. Nexus 7 devices running Kitkat have a kernel which often output an
+ *      incorrect 'idle' field in /proc/stat.  The value is close to twice the
+ *      correct value, and then returns to back to correct reading.  Both when
+ *      jumping up and back down we might create faulty CPU load readings.
+ */
+class CpuMonitor {
+  private static final String TAG = "CpuMonitor";
+  private static final int MOVING_AVERAGE_SAMPLES = 5;
+
+  private static final int CPU_STAT_SAMPLE_PERIOD_MS = 2000;
+  private static final int CPU_STAT_LOG_PERIOD_MS = 6000;
+
+  private final Context appContext;
+  // User CPU usage at current frequency.
+  private final MovingAverage userCpuUsage;
+  // System CPU usage at current frequency.
+  private final MovingAverage systemCpuUsage;
+  // Total CPU usage relative to maximum frequency.
+  private final MovingAverage totalCpuUsage;
+  // CPU frequency in percentage from maximum.
+  private final MovingAverage frequencyScale;
+
+  @Nullable
+  private ScheduledExecutorService executor;
+  private long lastStatLogTimeMs;
+  private long[] cpuFreqMax;
+  private int cpusPresent;
+  private int actualCpusPresent;
+  private boolean initialized;
+  private boolean cpuOveruse;
+  private String[] maxPath;
+  private String[] curPath;
+  private double[] curFreqScales;
+  @Nullable
+  private ProcStat lastProcStat;
+
+  private static class ProcStat {
+    final long userTime;
+    final long systemTime;
+    final long idleTime;
+
+    ProcStat(long userTime, long systemTime, long idleTime) {
+      this.userTime = userTime;
+      this.systemTime = systemTime;
+      this.idleTime = idleTime;
+    }
+  }
+
+  private static class MovingAverage {
+    private final int size;
+    private double sum;
+    private double currentValue;
+    private double[] circBuffer;
+    private int circBufferIndex;
+
+    public MovingAverage(int size) {
+      if (size <= 0) {
+        throw new AssertionError("Size value in MovingAverage ctor should be positive.");
+      }
+      this.size = size;
+      circBuffer = new double[size];
+    }
+
+    public void reset() {
+      Arrays.fill(circBuffer, 0);
+      circBufferIndex = 0;
+      sum = 0;
+      currentValue = 0;
+    }
+
+    public void addValue(double value) {
+      sum -= circBuffer[circBufferIndex];
+      circBuffer[circBufferIndex++] = value;
+      currentValue = value;
+      sum += value;
+      if (circBufferIndex >= size) {
+        circBufferIndex = 0;
+      }
+    }
+
+    public double getCurrent() {
+      return currentValue;
+    }
+
+    public double getAverage() {
+      return sum / (double) size;
+    }
+  }
+
+  public static boolean isSupported() {
+    return Build.VERSION.SDK_INT < Build.VERSION_CODES.N;
+  }
+
+  public CpuMonitor(Context context) {
+    if (!isSupported()) {
+      throw new RuntimeException("CpuMonitor is not supported on this Android version.");
+    }
+
+    Log.d(TAG, "CpuMonitor ctor.");
+    appContext = context.getApplicationContext();
+    userCpuUsage = new MovingAverage(MOVING_AVERAGE_SAMPLES);
+    systemCpuUsage = new MovingAverage(MOVING_AVERAGE_SAMPLES);
+    totalCpuUsage = new MovingAverage(MOVING_AVERAGE_SAMPLES);
+    frequencyScale = new MovingAverage(MOVING_AVERAGE_SAMPLES);
+    lastStatLogTimeMs = SystemClock.elapsedRealtime();
+
+    scheduleCpuUtilizationTask();
+  }
+
+  public void pause() {
+    if (executor != null) {
+      Log.d(TAG, "pause");
+      executor.shutdownNow();
+      executor = null;
+    }
+  }
+
+  public void resume() {
+    Log.d(TAG, "resume");
+    resetStat();
+    scheduleCpuUtilizationTask();
+  }
+
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public synchronized void reset() {
+    if (executor != null) {
+      Log.d(TAG, "reset");
+      resetStat();
+      cpuOveruse = false;
+    }
+  }
+
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public synchronized int getCpuUsageCurrent() {
+    return doubleToPercent(userCpuUsage.getCurrent() + systemCpuUsage.getCurrent());
+  }
+
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public synchronized int getCpuUsageAverage() {
+    return doubleToPercent(userCpuUsage.getAverage() + systemCpuUsage.getAverage());
+  }
+
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public synchronized int getFrequencyScaleAverage() {
+    return doubleToPercent(frequencyScale.getAverage());
+  }
+
+  private void scheduleCpuUtilizationTask() {
+    if (executor != null) {
+      executor.shutdownNow();
+      executor = null;
+    }
+
+    executor = Executors.newSingleThreadScheduledExecutor();
+    @SuppressWarnings("unused") // Prevent downstream linter warnings.
+    Future<?> possiblyIgnoredError = executor.scheduleAtFixedRate(new Runnable() {
+      @Override
+      public void run() {
+        cpuUtilizationTask();
+      }
+    }, 0, CPU_STAT_SAMPLE_PERIOD_MS, TimeUnit.MILLISECONDS);
+  }
+
+  private void cpuUtilizationTask() {
+    boolean cpuMonitorAvailable = sampleCpuUtilization();
+    if (cpuMonitorAvailable
+        && SystemClock.elapsedRealtime() - lastStatLogTimeMs >= CPU_STAT_LOG_PERIOD_MS) {
+      lastStatLogTimeMs = SystemClock.elapsedRealtime();
+      String statString = getStatString();
+      Log.d(TAG, statString);
+    }
+  }
+
+  private void init() {
+    try (FileInputStream fin = new FileInputStream("/sys/devices/system/cpu/present");
+         InputStreamReader streamReader = new InputStreamReader(fin, Charset.forName("UTF-8"));
+         BufferedReader reader = new BufferedReader(streamReader);
+         Scanner scanner = new Scanner(reader).useDelimiter("[-\n]");) {
+      scanner.nextInt(); // Skip leading number 0.
+      cpusPresent = 1 + scanner.nextInt();
+      scanner.close();
+    } catch (FileNotFoundException e) {
+      Log.e(TAG, "Cannot do CPU stats since /sys/devices/system/cpu/present is missing");
+    } catch (IOException e) {
+      Log.e(TAG, "Error closing file");
+    } catch (Exception e) {
+      Log.e(TAG, "Cannot do CPU stats due to /sys/devices/system/cpu/present parsing problem");
+    }
+
+    cpuFreqMax = new long[cpusPresent];
+    maxPath = new String[cpusPresent];
+    curPath = new String[cpusPresent];
+    curFreqScales = new double[cpusPresent];
+    for (int i = 0; i < cpusPresent; i++) {
+      cpuFreqMax[i] = 0; // Frequency "not yet determined".
+      curFreqScales[i] = 0;
+      maxPath[i] = "/sys/devices/system/cpu/cpu" + i + "/cpufreq/cpuinfo_max_freq";
+      curPath[i] = "/sys/devices/system/cpu/cpu" + i + "/cpufreq/scaling_cur_freq";
+    }
+
+    lastProcStat = new ProcStat(0, 0, 0);
+    resetStat();
+
+    initialized = true;
+  }
+
+  private synchronized void resetStat() {
+    userCpuUsage.reset();
+    systemCpuUsage.reset();
+    totalCpuUsage.reset();
+    frequencyScale.reset();
+    lastStatLogTimeMs = SystemClock.elapsedRealtime();
+  }
+
+  private int getBatteryLevel() {
+    // Use sticky broadcast with null receiver to read battery level once only.
+    Intent intent = appContext.registerReceiver(
+        null /* receiver */, new IntentFilter(Intent.ACTION_BATTERY_CHANGED));
+
+    int batteryLevel = 0;
+    int batteryScale = intent.getIntExtra(BatteryManager.EXTRA_SCALE, 100);
+    if (batteryScale > 0) {
+      batteryLevel =
+          (int) (100f * intent.getIntExtra(BatteryManager.EXTRA_LEVEL, 0) / batteryScale);
+    }
+    return batteryLevel;
+  }
+
+  /**
+   * Re-measure CPU use.  Call this method at an interval of around 1/s.
+   * This method returns true on success.  The fields
+   * cpuCurrent, cpuAvg3, and cpuAvgAll are updated on success, and represents:
+   * cpuCurrent: The CPU use since the last sampleCpuUtilization call.
+   * cpuAvg3: The average CPU over the last 3 calls.
+   * cpuAvgAll: The average CPU over the last SAMPLE_SAVE_NUMBER calls.
+   */
+  private synchronized boolean sampleCpuUtilization() {
+    long lastSeenMaxFreq = 0;
+    long cpuFreqCurSum = 0;
+    long cpuFreqMaxSum = 0;
+
+    if (!initialized) {
+      init();
+    }
+    if (cpusPresent == 0) {
+      return false;
+    }
+
+    actualCpusPresent = 0;
+    for (int i = 0; i < cpusPresent; i++) {
+      /*
+       * For each CPU, attempt to first read its max frequency, then its
+       * current frequency.  Once as the max frequency for a CPU is found,
+       * save it in cpuFreqMax[].
+       */
+
+      curFreqScales[i] = 0;
+      if (cpuFreqMax[i] == 0) {
+        // We have never found this CPU's max frequency.  Attempt to read it.
+        long cpufreqMax = readFreqFromFile(maxPath[i]);
+        if (cpufreqMax > 0) {
+          Log.d(TAG, "Core " + i + ". Max frequency: " + cpufreqMax);
+          lastSeenMaxFreq = cpufreqMax;
+          cpuFreqMax[i] = cpufreqMax;
+          maxPath[i] = null; // Kill path to free its memory.
+        }
+      } else {
+        lastSeenMaxFreq = cpuFreqMax[i]; // A valid, previously read value.
+      }
+
+      long cpuFreqCur = readFreqFromFile(curPath[i]);
+      if (cpuFreqCur == 0 && lastSeenMaxFreq == 0) {
+        // No current frequency information for this CPU core - ignore it.
+        continue;
+      }
+      if (cpuFreqCur > 0) {
+        actualCpusPresent++;
+      }
+      cpuFreqCurSum += cpuFreqCur;
+
+      /* Here, lastSeenMaxFreq might come from
+       * 1. cpuFreq[i], or
+       * 2. a previous iteration, or
+       * 3. a newly read value, or
+       * 4. hypothetically from the pre-loop dummy.
+       */
+      cpuFreqMaxSum += lastSeenMaxFreq;
+      if (lastSeenMaxFreq > 0) {
+        curFreqScales[i] = (double) cpuFreqCur / lastSeenMaxFreq;
+      }
+    }
+
+    if (cpuFreqCurSum == 0 || cpuFreqMaxSum == 0) {
+      Log.e(TAG, "Could not read max or current frequency for any CPU");
+      return false;
+    }
+
+    /*
+     * Since the cycle counts are for the period between the last invocation
+     * and this present one, we average the percentual CPU frequencies between
+     * now and the beginning of the measurement period.  This is significantly
+     * incorrect only if the frequencies have peeked or dropped in between the
+     * invocations.
+     */
+    double currentFrequencyScale = cpuFreqCurSum / (double) cpuFreqMaxSum;
+    if (frequencyScale.getCurrent() > 0) {
+      currentFrequencyScale = (frequencyScale.getCurrent() + currentFrequencyScale) * 0.5;
+    }
+
+    ProcStat procStat = readProcStat();
+    if (procStat == null) {
+      return false;
+    }
+
+    long diffUserTime = procStat.userTime - lastProcStat.userTime;
+    long diffSystemTime = procStat.systemTime - lastProcStat.systemTime;
+    long diffIdleTime = procStat.idleTime - lastProcStat.idleTime;
+    long allTime = diffUserTime + diffSystemTime + diffIdleTime;
+
+    if (currentFrequencyScale == 0 || allTime == 0) {
+      return false;
+    }
+
+    // Update statistics.
+    frequencyScale.addValue(currentFrequencyScale);
+
+    double currentUserCpuUsage = diffUserTime / (double) allTime;
+    userCpuUsage.addValue(currentUserCpuUsage);
+
+    double currentSystemCpuUsage = diffSystemTime / (double) allTime;
+    systemCpuUsage.addValue(currentSystemCpuUsage);
+
+    double currentTotalCpuUsage =
+        (currentUserCpuUsage + currentSystemCpuUsage) * currentFrequencyScale;
+    totalCpuUsage.addValue(currentTotalCpuUsage);
+
+    // Save new measurements for next round's deltas.
+    lastProcStat = procStat;
+
+    return true;
+  }
+
+  private int doubleToPercent(double d) {
+    return (int) (d * 100 + 0.5);
+  }
+
+  private synchronized String getStatString() {
+    StringBuilder stat = new StringBuilder();
+    stat.append("CPU User: ")
+        .append(doubleToPercent(userCpuUsage.getCurrent()))
+        .append("/")
+        .append(doubleToPercent(userCpuUsage.getAverage()))
+        .append(". System: ")
+        .append(doubleToPercent(systemCpuUsage.getCurrent()))
+        .append("/")
+        .append(doubleToPercent(systemCpuUsage.getAverage()))
+        .append(". Freq: ")
+        .append(doubleToPercent(frequencyScale.getCurrent()))
+        .append("/")
+        .append(doubleToPercent(frequencyScale.getAverage()))
+        .append(". Total usage: ")
+        .append(doubleToPercent(totalCpuUsage.getCurrent()))
+        .append("/")
+        .append(doubleToPercent(totalCpuUsage.getAverage()))
+        .append(". Cores: ")
+        .append(actualCpusPresent);
+    stat.append("( ");
+    for (int i = 0; i < cpusPresent; i++) {
+      stat.append(doubleToPercent(curFreqScales[i])).append(" ");
+    }
+    stat.append("). Battery: ").append(getBatteryLevel());
+    if (cpuOveruse) {
+      stat.append(". Overuse.");
+    }
+    return stat.toString();
+  }
+
+  /**
+   * Read a single integer value from the named file.  Return the read value
+   * or if an error occurs return 0.
+   */
+  private long readFreqFromFile(String fileName) {
+    long number = 0;
+    try (FileInputStream stream = new FileInputStream(fileName);
+         InputStreamReader streamReader = new InputStreamReader(stream, Charset.forName("UTF-8"));
+         BufferedReader reader = new BufferedReader(streamReader)) {
+      String line = reader.readLine();
+      number = parseLong(line);
+    } catch (FileNotFoundException e) {
+      // CPU core is off, so file with its scaling frequency .../cpufreq/scaling_cur_freq
+      // is not present. This is not an error.
+    } catch (IOException e) {
+      // CPU core is off, so file with its scaling frequency .../cpufreq/scaling_cur_freq
+      // is empty. This is not an error.
+    }
+    return number;
+  }
+
+  private static long parseLong(String value) {
+    long number = 0;
+    try {
+      number = Long.parseLong(value);
+    } catch (NumberFormatException e) {
+      Log.e(TAG, "parseLong error.", e);
+    }
+    return number;
+  }
+
+  /*
+   * Read the current utilization of all CPUs using the cumulative first line
+   * of /proc/stat.
+   */
+  @SuppressWarnings("StringSplitter")
+  private @Nullable ProcStat readProcStat() {
+    long userTime = 0;
+    long systemTime = 0;
+    long idleTime = 0;
+    try (FileInputStream stream = new FileInputStream("/proc/stat");
+         InputStreamReader streamReader = new InputStreamReader(stream, Charset.forName("UTF-8"));
+         BufferedReader reader = new BufferedReader(streamReader)) {
+      // line should contain something like this:
+      // cpu  5093818 271838 3512830 165934119 101374 447076 272086 0 0 0
+      //       user    nice  system     idle   iowait  irq   softirq
+      String line = reader.readLine();
+      String[] lines = line.split("\\s+");
+      int length = lines.length;
+      if (length >= 5) {
+        userTime = parseLong(lines[1]); // user
+        userTime += parseLong(lines[2]); // nice
+        systemTime = parseLong(lines[3]); // system
+        idleTime = parseLong(lines[4]); // idle
+      }
+      if (length >= 8) {
+        userTime += parseLong(lines[5]); // iowait
+        systemTime += parseLong(lines[6]); // irq
+        systemTime += parseLong(lines[7]); // softirq
+      }
+    } catch (FileNotFoundException e) {
+      Log.e(TAG, "Cannot open /proc/stat for reading", e);
+      return null;
+    } catch (Exception e) {
+      Log.e(TAG, "Problems parsing /proc/stat", e);
+      return null;
+    }
+    return new ProcStat(userTime, systemTime, idleTime);
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/DirectRTCClient.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/DirectRTCClient.java
new file mode 100644
index 0000000000..1b113e1398
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/DirectRTCClient.java
@@ -0,0 +1,346 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.util.Log;
+import androidx.annotation.Nullable;
+import java.util.ArrayList;
+import java.util.concurrent.ExecutorService;
+import java.util.concurrent.Executors;
+import java.util.regex.Matcher;
+import java.util.regex.Pattern;
+import org.json.JSONArray;
+import org.json.JSONException;
+import org.json.JSONObject;
+import org.webrtc.IceCandidate;
+import org.webrtc.SessionDescription;
+
+/**
+ * Implementation of AppRTCClient that uses direct TCP connection as the signaling channel.
+ * This eliminates the need for an external server. This class does not support loopback
+ * connections.
+ */
+public class DirectRTCClient implements AppRTCClient, TCPChannelClient.TCPChannelEvents {
+  private static final String TAG = "DirectRTCClient";
+  private static final int DEFAULT_PORT = 8888;
+
+  // Regex pattern used for checking if room id looks like an IP.
+  static final Pattern IP_PATTERN = Pattern.compile("("
+      // IPv4
+      + "((\\d+\\.){3}\\d+)|"
+      // IPv6
+      + "\\[((([0-9a-fA-F]{1,4}:)*[0-9a-fA-F]{1,4})?::"
+      + "(([0-9a-fA-F]{1,4}:)*[0-9a-fA-F]{1,4})?)\\]|"
+      + "\\[(([0-9a-fA-F]{1,4}:){7}[0-9a-fA-F]{1,4})\\]|"
+      // IPv6 without []
+      + "((([0-9a-fA-F]{1,4}:)*[0-9a-fA-F]{1,4})?::(([0-9a-fA-F]{1,4}:)*[0-9a-fA-F]{1,4})?)|"
+      + "(([0-9a-fA-F]{1,4}:){7}[0-9a-fA-F]{1,4})|"
+      // Literals
+      + "localhost"
+      + ")"
+      // Optional port number
+      + "(:(\\d+))?");
+
+  private final ExecutorService executor;
+  private final SignalingEvents events;
+  @Nullable
+  private TCPChannelClient tcpClient;
+  private RoomConnectionParameters connectionParameters;
+
+  private enum ConnectionState { NEW, CONNECTED, CLOSED, ERROR }
+
+  // All alterations of the room state should be done from inside the looper thread.
+  private ConnectionState roomState;
+
+  public DirectRTCClient(SignalingEvents events) {
+    this.events = events;
+
+    executor = Executors.newSingleThreadExecutor();
+    roomState = ConnectionState.NEW;
+  }
+
+  /**
+   * Connects to the room, roomId in connectionsParameters is required. roomId must be a valid
+   * IP address matching IP_PATTERN.
+   */
+  @Override
+  public void connectToRoom(RoomConnectionParameters connectionParameters) {
+    this.connectionParameters = connectionParameters;
+
+    if (connectionParameters.loopback) {
+      reportError("Loopback connections aren't supported by DirectRTCClient.");
+    }
+
+    executor.execute(new Runnable() {
+      @Override
+      public void run() {
+        connectToRoomInternal();
+      }
+    });
+  }
+
+  @Override
+  public void disconnectFromRoom() {
+    executor.execute(new Runnable() {
+      @Override
+      public void run() {
+        disconnectFromRoomInternal();
+      }
+    });
+  }
+
+  /**
+   * Connects to the room.
+   *
+   * Runs on the looper thread.
+   */
+  private void connectToRoomInternal() {
+    this.roomState = ConnectionState.NEW;
+
+    String endpoint = connectionParameters.roomId;
+
+    Matcher matcher = IP_PATTERN.matcher(endpoint);
+    if (!matcher.matches()) {
+      reportError("roomId must match IP_PATTERN for DirectRTCClient.");
+      return;
+    }
+
+    String ip = matcher.group(1);
+    String portStr = matcher.group(matcher.groupCount());
+    int port;
+
+    if (portStr != null) {
+      try {
+        port = Integer.parseInt(portStr);
+      } catch (NumberFormatException e) {
+        reportError("Invalid port number: " + portStr);
+        return;
+      }
+    } else {
+      port = DEFAULT_PORT;
+    }
+
+    tcpClient = new TCPChannelClient(executor, this, ip, port);
+  }
+
+  /**
+   * Disconnects from the room.
+   *
+   * Runs on the looper thread.
+   */
+  private void disconnectFromRoomInternal() {
+    roomState = ConnectionState.CLOSED;
+
+    if (tcpClient != null) {
+      tcpClient.disconnect();
+      tcpClient = null;
+    }
+    executor.shutdown();
+  }
+
+  @Override
+  public void sendOfferSdp(final SessionDescription sdp) {
+    executor.execute(new Runnable() {
+      @Override
+      public void run() {
+        if (roomState != ConnectionState.CONNECTED) {
+          reportError("Sending offer SDP in non connected state.");
+          return;
+        }
+        JSONObject json = new JSONObject();
+        jsonPut(json, "sdp", sdp.description);
+        jsonPut(json, "type", "offer");
+        sendMessage(json.toString());
+      }
+    });
+  }
+
+  @Override
+  public void sendAnswerSdp(final SessionDescription sdp) {
+    executor.execute(new Runnable() {
+      @Override
+      public void run() {
+        JSONObject json = new JSONObject();
+        jsonPut(json, "sdp", sdp.description);
+        jsonPut(json, "type", "answer");
+        sendMessage(json.toString());
+      }
+    });
+  }
+
+  @Override
+  public void sendLocalIceCandidate(final IceCandidate candidate) {
+    executor.execute(new Runnable() {
+      @Override
+      public void run() {
+        JSONObject json = new JSONObject();
+        jsonPut(json, "type", "candidate");
+        jsonPut(json, "label", candidate.sdpMLineIndex);
+        jsonPut(json, "id", candidate.sdpMid);
+        jsonPut(json, "candidate", candidate.sdp);
+
+        if (roomState != ConnectionState.CONNECTED) {
+          reportError("Sending ICE candidate in non connected state.");
+          return;
+        }
+        sendMessage(json.toString());
+      }
+    });
+  }
+
+  /** Send removed Ice candidates to the other participant. */
+  @Override
+  public void sendLocalIceCandidateRemovals(final IceCandidate[] candidates) {
+    executor.execute(new Runnable() {
+      @Override
+      public void run() {
+        JSONObject json = new JSONObject();
+        jsonPut(json, "type", "remove-candidates");
+        JSONArray jsonArray = new JSONArray();
+        for (final IceCandidate candidate : candidates) {
+          jsonArray.put(toJsonCandidate(candidate));
+        }
+        jsonPut(json, "candidates", jsonArray);
+
+        if (roomState != ConnectionState.CONNECTED) {
+          reportError("Sending ICE candidate removals in non connected state.");
+          return;
+        }
+        sendMessage(json.toString());
+      }
+    });
+  }
+
+  // -------------------------------------------------------------------
+  // TCPChannelClient event handlers
+
+  /**
+   * If the client is the server side, this will trigger onConnectedToRoom.
+   */
+  @Override
+  public void onTCPConnected(boolean isServer) {
+    if (isServer) {
+      roomState = ConnectionState.CONNECTED;
+
+      SignalingParameters parameters = new SignalingParameters(
+          // Ice servers are not needed for direct connections.
+          new ArrayList<>(),
+          isServer, // Server side acts as the initiator on direct connections.
+          null, // clientId
+          null, // wssUrl
+          null, // wwsPostUrl
+          null, // offerSdp
+          null // iceCandidates
+          );
+      events.onConnectedToRoom(parameters);
+    }
+  }
+
+  @Override
+  public void onTCPMessage(String msg) {
+    try {
+      JSONObject json = new JSONObject(msg);
+      String type = json.optString("type");
+      if (type.equals("candidate")) {
+        events.onRemoteIceCandidate(toJavaCandidate(json));
+      } else if (type.equals("remove-candidates")) {
+        JSONArray candidateArray = json.getJSONArray("candidates");
+        IceCandidate[] candidates = new IceCandidate[candidateArray.length()];
+        for (int i = 0; i < candidateArray.length(); ++i) {
+          candidates[i] = toJavaCandidate(candidateArray.getJSONObject(i));
+        }
+        events.onRemoteIceCandidatesRemoved(candidates);
+      } else if (type.equals("answer")) {
+        SessionDescription sdp = new SessionDescription(
+            SessionDescription.Type.fromCanonicalForm(type), json.getString("sdp"));
+        events.onRemoteDescription(sdp);
+      } else if (type.equals("offer")) {
+        SessionDescription sdp = new SessionDescription(
+            SessionDescription.Type.fromCanonicalForm(type), json.getString("sdp"));
+
+        SignalingParameters parameters = new SignalingParameters(
+            // Ice servers are not needed for direct connections.
+            new ArrayList<>(),
+            false, // This code will only be run on the client side. So, we are not the initiator.
+            null, // clientId
+            null, // wssUrl
+            null, // wssPostUrl
+            sdp, // offerSdp
+            null // iceCandidates
+            );
+        roomState = ConnectionState.CONNECTED;
+        events.onConnectedToRoom(parameters);
+      } else {
+        reportError("Unexpected TCP message: " + msg);
+      }
+    } catch (JSONException e) {
+      reportError("TCP message JSON parsing error: " + e.toString());
+    }
+  }
+
+  @Override
+  public void onTCPError(String description) {
+    reportError("TCP connection error: " + description);
+  }
+
+  @Override
+  public void onTCPClose() {
+    events.onChannelClose();
+  }
+
+  // --------------------------------------------------------------------
+  // Helper functions.
+  private void reportError(final String errorMessage) {
+    Log.e(TAG, errorMessage);
+    executor.execute(new Runnable() {
+      @Override
+      public void run() {
+        if (roomState != ConnectionState.ERROR) {
+          roomState = ConnectionState.ERROR;
+          events.onChannelError(errorMessage);
+        }
+      }
+    });
+  }
+
+  private void sendMessage(final String message) {
+    executor.execute(new Runnable() {
+      @Override
+      public void run() {
+        tcpClient.send(message);
+      }
+    });
+  }
+
+  // Put a `key`->`value` mapping in `json`.
+  private static void jsonPut(JSONObject json, String key, Object value) {
+    try {
+      json.put(key, value);
+    } catch (JSONException e) {
+      throw new RuntimeException(e);
+    }
+  }
+
+  // Converts a Java candidate to a JSONObject.
+  private static JSONObject toJsonCandidate(final IceCandidate candidate) {
+    JSONObject json = new JSONObject();
+    jsonPut(json, "label", candidate.sdpMLineIndex);
+    jsonPut(json, "id", candidate.sdpMid);
+    jsonPut(json, "candidate", candidate.sdp);
+    return json;
+  }
+
+  // Converts a JSON candidate to a Java object.
+  private static IceCandidate toJavaCandidate(JSONObject json) throws JSONException {
+    return new IceCandidate(
+        json.getString("id"), json.getInt("label"), json.getString("candidate"));
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/HudFragment.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/HudFragment.java
new file mode 100644
index 0000000000..94ca05549a
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/HudFragment.java
@@ -0,0 +1,102 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.app.Fragment;
+import android.os.Bundle;
+import android.view.LayoutInflater;
+import android.view.View;
+import android.view.ViewGroup;
+import android.widget.ImageButton;
+import android.widget.TextView;
+import org.webrtc.RTCStats;
+import org.webrtc.RTCStatsReport;
+
+/**
+ * Fragment for HUD statistics display.
+ */
+public class HudFragment extends Fragment {
+  private TextView statView;
+  private ImageButton toggleDebugButton;
+  private boolean displayHud;
+  private volatile boolean isRunning;
+  private CpuMonitor cpuMonitor;
+
+  @Override
+  public View onCreateView(
+      LayoutInflater inflater, ViewGroup container, Bundle savedInstanceState) {
+    View controlView = inflater.inflate(R.layout.fragment_hud, container, false);
+
+    // Create UI controls.
+    statView = controlView.findViewById(R.id.hud_stat_call);
+    toggleDebugButton = controlView.findViewById(R.id.button_toggle_debug);
+
+    toggleDebugButton.setOnClickListener(new View.OnClickListener() {
+      @Override
+      public void onClick(View view) {
+        if (displayHud) {
+          statView.setVisibility(
+              statView.getVisibility() == View.VISIBLE ? View.INVISIBLE : View.VISIBLE);
+        }
+      }
+    });
+
+    return controlView;
+  }
+
+  @Override
+  public void onStart() {
+    super.onStart();
+
+    Bundle args = getArguments();
+    if (args != null) {
+      displayHud = args.getBoolean(CallActivity.EXTRA_DISPLAY_HUD, false);
+    }
+    int visibility = displayHud ? View.VISIBLE : View.INVISIBLE;
+    statView.setVisibility(View.INVISIBLE);
+    toggleDebugButton.setVisibility(visibility);
+    isRunning = true;
+  }
+
+  @Override
+  public void onStop() {
+    isRunning = false;
+    super.onStop();
+  }
+
+  public void setCpuMonitor(CpuMonitor cpuMonitor) {
+    this.cpuMonitor = cpuMonitor;
+  }
+
+  public void updateEncoderStatistics(final RTCStatsReport report) {
+    if (!isRunning || !displayHud) {
+      return;
+    }
+
+    StringBuilder sb = new StringBuilder();
+
+    if (cpuMonitor != null) {
+      sb.append("CPU%: ")
+          .append(cpuMonitor.getCpuUsageCurrent())
+          .append("/")
+          .append(cpuMonitor.getCpuUsageAverage())
+          .append(". Freq: ")
+          .append(cpuMonitor.getFrequencyScaleAverage())
+          .append("\n");
+    }
+
+    for (RTCStats stat : report.getStatsMap().values()) {
+      sb.append(stat.toString()).append("\n");
+    }
+
+    statView.setText(sb.toString());
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
new file mode 100644
index 0000000000..7bdce00b2f
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
@@ -0,0 +1,1402 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.content.Context;
+import android.os.Environment;
+import android.os.ParcelFileDescriptor;
+import android.util.Log;
+import androidx.annotation.Nullable;
+import java.io.File;
+import java.io.IOException;
+import java.nio.ByteBuffer;
+import java.nio.charset.Charset;
+import java.text.DateFormat;
+import java.text.SimpleDateFormat;
+import java.util.ArrayList;
+import java.util.Arrays;
+import java.util.Collections;
+import java.util.Date;
+import java.util.Iterator;
+import java.util.List;
+import java.util.Locale;
+import java.util.Timer;
+import java.util.TimerTask;
+import java.util.concurrent.ExecutorService;
+import java.util.concurrent.Executors;
+import java.util.regex.Matcher;
+import java.util.regex.Pattern;
+import org.appspot.apprtc.AppRTCClient.SignalingParameters;
+import org.appspot.apprtc.RecordedAudioToFileController;
+import org.webrtc.AddIceObserver;
+import org.webrtc.AudioSource;
+import org.webrtc.AudioTrack;
+import org.webrtc.CameraVideoCapturer;
+import org.webrtc.CandidatePairChangeEvent;
+import org.webrtc.DataChannel;
+import org.webrtc.DefaultVideoDecoderFactory;
+import org.webrtc.DefaultVideoEncoderFactory;
+import org.webrtc.EglBase;
+import org.webrtc.IceCandidate;
+import org.webrtc.IceCandidateErrorEvent;
+import org.webrtc.Logging;
+import org.webrtc.MediaConstraints;
+import org.webrtc.MediaStream;
+import org.webrtc.MediaStreamTrack;
+import org.webrtc.PeerConnection;
+import org.webrtc.PeerConnection.IceConnectionState;
+import org.webrtc.PeerConnection.PeerConnectionState;
+import org.webrtc.PeerConnectionFactory;
+import org.webrtc.RTCStatsCollectorCallback;
+import org.webrtc.RTCStatsReport;
+import org.webrtc.RtpParameters;
+import org.webrtc.RtpReceiver;
+import org.webrtc.RtpSender;
+import org.webrtc.RtpTransceiver;
+import org.webrtc.SdpObserver;
+import org.webrtc.SessionDescription;
+import org.webrtc.SoftwareVideoDecoderFactory;
+import org.webrtc.SoftwareVideoEncoderFactory;
+import org.webrtc.SurfaceTextureHelper;
+import org.webrtc.VideoCapturer;
+import org.webrtc.VideoDecoderFactory;
+import org.webrtc.VideoEncoderFactory;
+import org.webrtc.VideoSink;
+import org.webrtc.VideoSource;
+import org.webrtc.VideoTrack;
+import org.webrtc.audio.AudioDeviceModule;
+import org.webrtc.audio.JavaAudioDeviceModule;
+import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordErrorCallback;
+import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordStateCallback;
+import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackErrorCallback;
+import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackStateCallback;
+
+/**
+ * Peer connection client implementation.
+ *
+ * <p>All public methods are routed to local looper thread.
+ * All PeerConnectionEvents callbacks are invoked from the same looper thread.
+ * This class is a singleton.
+ */
+public class PeerConnectionClient {
+  public static final String VIDEO_TRACK_ID = "ARDAMSv0";
+  public static final String AUDIO_TRACK_ID = "ARDAMSa0";
+  public static final String VIDEO_TRACK_TYPE = "video";
+  private static final String TAG = "PCRTCClient";
+  private static final String VIDEO_CODEC_VP8 = "VP8";
+  private static final String VIDEO_CODEC_VP9 = "VP9";
+  private static final String VIDEO_CODEC_H264 = "H264";
+  private static final String VIDEO_CODEC_H264_BASELINE = "H264 Baseline";
+  private static final String VIDEO_CODEC_H264_HIGH = "H264 High";
+  private static final String VIDEO_CODEC_AV1 = "AV1";
+  private static final String AUDIO_CODEC_OPUS = "opus";
+  private static final String AUDIO_CODEC_ISAC = "ISAC";
+  private static final String VIDEO_CODEC_PARAM_START_BITRATE = "x-google-start-bitrate";
+  private static final String VIDEO_FLEXFEC_FIELDTRIAL =
+      "WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/";
+  private static final String VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL = "WebRTC-IntelVP8/Enabled/";
+  private static final String DISABLE_WEBRTC_AGC_FIELDTRIAL =
+      "WebRTC-Audio-MinimizeResamplingOnMobile/Enabled/";
+  private static final String AUDIO_CODEC_PARAM_BITRATE = "maxaveragebitrate";
+  private static final String AUDIO_ECHO_CANCELLATION_CONSTRAINT = "googEchoCancellation";
+  private static final String AUDIO_AUTO_GAIN_CONTROL_CONSTRAINT = "googAutoGainControl";
+  private static final String AUDIO_HIGH_PASS_FILTER_CONSTRAINT = "googHighpassFilter";
+  private static final String AUDIO_NOISE_SUPPRESSION_CONSTRAINT = "googNoiseSuppression";
+  private static final String DTLS_SRTP_KEY_AGREEMENT_CONSTRAINT = "DtlsSrtpKeyAgreement";
+  private static final int HD_VIDEO_WIDTH = 1280;
+  private static final int HD_VIDEO_HEIGHT = 720;
+  private static final int BPS_IN_KBPS = 1000;
+  private static final String RTCEVENTLOG_OUTPUT_DIR_NAME = "rtc_event_log";
+
+  // Executor thread is started once in private ctor and is used for all
+  // peer connection API calls to ensure new peer connection factory is
+  // created on the same thread as previously destroyed factory.
+  private static final ExecutorService executor = Executors.newSingleThreadExecutor();
+
+  private final PCObserver pcObserver = new PCObserver();
+  private final SDPObserver sdpObserver = new SDPObserver();
+  private final Timer statsTimer = new Timer();
+  private final EglBase rootEglBase;
+  private final Context appContext;
+  private final PeerConnectionParameters peerConnectionParameters;
+  private final PeerConnectionEvents events;
+
+  @Nullable
+  private PeerConnectionFactory factory;
+  @Nullable
+  private PeerConnection peerConnection;
+  @Nullable
+  private AudioSource audioSource;
+  @Nullable private SurfaceTextureHelper surfaceTextureHelper;
+  @Nullable private VideoSource videoSource;
+  private boolean preferIsac;
+  private boolean videoCapturerStopped;
+  private boolean isError;
+  @Nullable
+  private VideoSink localRender;
+  @Nullable private List<VideoSink> remoteSinks;
+  private SignalingParameters signalingParameters;
+  private int videoWidth;
+  private int videoHeight;
+  private int videoFps;
+  private MediaConstraints audioConstraints;
+  private MediaConstraints sdpMediaConstraints;
+  // Queued remote ICE candidates are consumed only after both local and
+  // remote descriptions are set. Similarly local ICE candidates are sent to
+  // remote peer after both local and remote description are set.
+  @Nullable
+  private List<IceCandidate> queuedRemoteCandidates;
+  private boolean isInitiator;
+  @Nullable private SessionDescription localDescription; // either offer or answer description
+  @Nullable
+  private VideoCapturer videoCapturer;
+  // enableVideo is set to true if video should be rendered and sent.
+  private boolean renderVideo = true;
+  @Nullable
+  private VideoTrack localVideoTrack;
+  @Nullable
+  private VideoTrack remoteVideoTrack;
+  @Nullable
+  private RtpSender localVideoSender;
+  // enableAudio is set to true if audio should be sent.
+  private boolean enableAudio = true;
+  @Nullable
+  private AudioTrack localAudioTrack;
+  @Nullable
+  private DataChannel dataChannel;
+  private final boolean dataChannelEnabled;
+  // Enable RtcEventLog.
+  @Nullable
+  private RtcEventLog rtcEventLog;
+  // Implements the WebRtcAudioRecordSamplesReadyCallback interface and writes
+  // recorded audio samples to an output file.
+  @Nullable private RecordedAudioToFileController saveRecordedAudioToFile;
+
+  /**
+   * Peer connection parameters.
+   */
+  public static class DataChannelParameters {
+    public final boolean ordered;
+    public final int maxRetransmitTimeMs;
+    public final int maxRetransmits;
+    public final String protocol;
+    public final boolean negotiated;
+    public final int id;
+
+    public DataChannelParameters(boolean ordered, int maxRetransmitTimeMs, int maxRetransmits,
+        String protocol, boolean negotiated, int id) {
+      this.ordered = ordered;
+      this.maxRetransmitTimeMs = maxRetransmitTimeMs;
+      this.maxRetransmits = maxRetransmits;
+      this.protocol = protocol;
+      this.negotiated = negotiated;
+      this.id = id;
+    }
+  }
+
+  /**
+   * Peer connection parameters.
+   */
+  public static class PeerConnectionParameters {
+    public final boolean videoCallEnabled;
+    public final boolean loopback;
+    public final boolean tracing;
+    public final int videoWidth;
+    public final int videoHeight;
+    public final int videoFps;
+    public final int videoMaxBitrate;
+    public final String videoCodec;
+    public final boolean videoCodecHwAcceleration;
+    public final boolean videoFlexfecEnabled;
+    public final int audioStartBitrate;
+    public final String audioCodec;
+    public final boolean noAudioProcessing;
+    public final boolean aecDump;
+    public final boolean saveInputAudioToFile;
+    public final boolean useOpenSLES;
+    public final boolean disableBuiltInAEC;
+    public final boolean disableBuiltInAGC;
+    public final boolean disableBuiltInNS;
+    public final boolean disableWebRtcAGCAndHPF;
+    public final boolean enableRtcEventLog;
+    private final DataChannelParameters dataChannelParameters;
+
+    public PeerConnectionParameters(boolean videoCallEnabled, boolean loopback, boolean tracing,
+        int videoWidth, int videoHeight, int videoFps, int videoMaxBitrate, String videoCodec,
+        boolean videoCodecHwAcceleration, boolean videoFlexfecEnabled, int audioStartBitrate,
+        String audioCodec, boolean noAudioProcessing, boolean aecDump, boolean saveInputAudioToFile,
+        boolean useOpenSLES, boolean disableBuiltInAEC, boolean disableBuiltInAGC,
+        boolean disableBuiltInNS, boolean disableWebRtcAGCAndHPF, boolean enableRtcEventLog,
+        DataChannelParameters dataChannelParameters) {
+      this.videoCallEnabled = videoCallEnabled;
+      this.loopback = loopback;
+      this.tracing = tracing;
+      this.videoWidth = videoWidth;
+      this.videoHeight = videoHeight;
+      this.videoFps = videoFps;
+      this.videoMaxBitrate = videoMaxBitrate;
+      this.videoCodec = videoCodec;
+      this.videoFlexfecEnabled = videoFlexfecEnabled;
+      this.videoCodecHwAcceleration = videoCodecHwAcceleration;
+      this.audioStartBitrate = audioStartBitrate;
+      this.audioCodec = audioCodec;
+      this.noAudioProcessing = noAudioProcessing;
+      this.aecDump = aecDump;
+      this.saveInputAudioToFile = saveInputAudioToFile;
+      this.useOpenSLES = useOpenSLES;
+      this.disableBuiltInAEC = disableBuiltInAEC;
+      this.disableBuiltInAGC = disableBuiltInAGC;
+      this.disableBuiltInNS = disableBuiltInNS;
+      this.disableWebRtcAGCAndHPF = disableWebRtcAGCAndHPF;
+      this.enableRtcEventLog = enableRtcEventLog;
+      this.dataChannelParameters = dataChannelParameters;
+    }
+  }
+
+  /**
+   * Peer connection events.
+   */
+  public interface PeerConnectionEvents {
+    /**
+     * Callback fired once local SDP is created and set.
+     */
+    void onLocalDescription(final SessionDescription sdp);
+
+    /**
+     * Callback fired once local Ice candidate is generated.
+     */
+    void onIceCandidate(final IceCandidate candidate);
+
+    /**
+     * Callback fired once local ICE candidates are removed.
+     */
+    void onIceCandidatesRemoved(final IceCandidate[] candidates);
+
+    /**
+     * Callback fired once connection is established (IceConnectionState is
+     * CONNECTED).
+     */
+    void onIceConnected();
+
+    /**
+     * Callback fired once connection is disconnected (IceConnectionState is
+     * DISCONNECTED).
+     */
+    void onIceDisconnected();
+
+    /**
+     * Callback fired once DTLS connection is established (PeerConnectionState
+     * is CONNECTED).
+     */
+    void onConnected();
+
+    /**
+     * Callback fired once DTLS connection is disconnected (PeerConnectionState
+     * is DISCONNECTED).
+     */
+    void onDisconnected();
+
+    /**
+     * Callback fired once peer connection is closed.
+     */
+    void onPeerConnectionClosed();
+
+    /**
+     * Callback fired once peer connection statistics is ready.
+     */
+    void onPeerConnectionStatsReady(final RTCStatsReport report);
+
+    /**
+     * Callback fired once peer connection error happened.
+     */
+    void onPeerConnectionError(final String description);
+  }
+
+  /**
+   * Create a PeerConnectionClient with the specified parameters. PeerConnectionClient takes
+   * ownership of `eglBase`.
+   */
+  public PeerConnectionClient(Context appContext, EglBase eglBase,
+      PeerConnectionParameters peerConnectionParameters, PeerConnectionEvents events) {
+    this.rootEglBase = eglBase;
+    this.appContext = appContext;
+    this.events = events;
+    this.peerConnectionParameters = peerConnectionParameters;
+    this.dataChannelEnabled = peerConnectionParameters.dataChannelParameters != null;
+
+    Log.d(TAG, "Preferred video codec: " + getSdpVideoCodecName(peerConnectionParameters));
+
+    final String fieldTrials = getFieldTrials(peerConnectionParameters);
+    executor.execute(() -> {
+      Log.d(TAG, "Initialize WebRTC. Field trials: " + fieldTrials);
+      PeerConnectionFactory.initialize(
+          PeerConnectionFactory.InitializationOptions.builder(appContext)
+              .setFieldTrials(fieldTrials)
+              .setEnableInternalTracer(true)
+              .createInitializationOptions());
+    });
+  }
+
+  /**
+   * This function should only be called once.
+   */
+  public void createPeerConnectionFactory(PeerConnectionFactory.Options options) {
+    if (factory != null) {
+      throw new IllegalStateException("PeerConnectionFactory has already been constructed");
+    }
+    executor.execute(() -> createPeerConnectionFactoryInternal(options));
+  }
+
+  public void createPeerConnection(final VideoSink localRender, final VideoSink remoteSink,
+      final VideoCapturer videoCapturer, final SignalingParameters signalingParameters) {
+    if (peerConnectionParameters.videoCallEnabled && videoCapturer == null) {
+      Log.w(TAG, "Video call enabled but no video capturer provided.");
+    }
+    createPeerConnection(
+        localRender, Collections.singletonList(remoteSink), videoCapturer, signalingParameters);
+  }
+
+  public void createPeerConnection(final VideoSink localRender, final List<VideoSink> remoteSinks,
+      final VideoCapturer videoCapturer, final SignalingParameters signalingParameters) {
+    if (peerConnectionParameters == null) {
+      Log.e(TAG, "Creating peer connection without initializing factory.");
+      return;
+    }
+    this.localRender = localRender;
+    this.remoteSinks = remoteSinks;
+    this.videoCapturer = videoCapturer;
+    this.signalingParameters = signalingParameters;
+    executor.execute(() -> {
+      try {
+        createMediaConstraintsInternal();
+        createPeerConnectionInternal();
+        maybeCreateAndStartRtcEventLog();
+      } catch (Exception e) {
+        reportError("Failed to create peer connection: " + e.getMessage());
+        throw e;
+      }
+    });
+  }
+
+  public void close() {
+    executor.execute(this ::closeInternal);
+  }
+
+  private boolean isVideoCallEnabled() {
+    return peerConnectionParameters.videoCallEnabled && videoCapturer != null;
+  }
+
+  private void createPeerConnectionFactoryInternal(PeerConnectionFactory.Options options) {
+    isError = false;
+
+    if (peerConnectionParameters.tracing) {
+      PeerConnectionFactory.startInternalTracingCapture(
+          Environment.getExternalStorageDirectory().getAbsolutePath() + File.separator
+          + "webrtc-trace.txt");
+    }
+
+    // Check if ISAC is used by default.
+    preferIsac = peerConnectionParameters.audioCodec != null
+        && peerConnectionParameters.audioCodec.equals(AUDIO_CODEC_ISAC);
+
+    // It is possible to save a copy in raw PCM format on a file by checking
+    // the "Save input audio to file" checkbox in the Settings UI. A callback
+    // interface is set when this flag is enabled. As a result, a copy of recorded
+    // audio samples are provided to this client directly from the native audio
+    // layer in Java.
+    if (peerConnectionParameters.saveInputAudioToFile) {
+      if (!peerConnectionParameters.useOpenSLES) {
+        Log.d(TAG, "Enable recording of microphone input audio to file");
+        saveRecordedAudioToFile = new RecordedAudioToFileController(executor);
+      } else {
+        // TODO(henrika): ensure that the UI reflects that if OpenSL ES is selected,
+        // then the "Save inut audio to file" option shall be grayed out.
+        Log.e(TAG, "Recording of input audio is not supported for OpenSL ES");
+      }
+    }
+
+    final AudioDeviceModule adm = createJavaAudioDevice();
+
+    // Create peer connection factory.
+    if (options != null) {
+      Log.d(TAG, "Factory networkIgnoreMask option: " + options.networkIgnoreMask);
+    }
+    final boolean enableH264HighProfile =
+        VIDEO_CODEC_H264_HIGH.equals(peerConnectionParameters.videoCodec);
+    final VideoEncoderFactory encoderFactory;
+    final VideoDecoderFactory decoderFactory;
+
+    if (peerConnectionParameters.videoCodecHwAcceleration) {
+      encoderFactory = new DefaultVideoEncoderFactory(
+          rootEglBase.getEglBaseContext(), true /* enableIntelVp8Encoder */, enableH264HighProfile);
+      decoderFactory = new DefaultVideoDecoderFactory(rootEglBase.getEglBaseContext());
+    } else {
+      encoderFactory = new SoftwareVideoEncoderFactory();
+      decoderFactory = new SoftwareVideoDecoderFactory();
+    }
+
+    // Disable encryption for loopback calls.
+    if (peerConnectionParameters.loopback) {
+      options.disableEncryption = true;
+    }
+    factory = PeerConnectionFactory.builder()
+                  .setOptions(options)
+                  .setAudioDeviceModule(adm)
+                  .setVideoEncoderFactory(encoderFactory)
+                  .setVideoDecoderFactory(decoderFactory)
+                  .createPeerConnectionFactory();
+    Log.d(TAG, "Peer connection factory created.");
+    adm.release();
+  }
+
+  AudioDeviceModule createJavaAudioDevice() {
+    // Enable/disable OpenSL ES playback.
+    if (!peerConnectionParameters.useOpenSLES) {
+      Log.w(TAG, "External OpenSLES ADM not implemented yet.");
+      // TODO(magjed): Add support for external OpenSLES ADM.
+    }
+
+    // Set audio record error callbacks.
+    AudioRecordErrorCallback audioRecordErrorCallback = new AudioRecordErrorCallback() {
+      @Override
+      public void onWebRtcAudioRecordInitError(String errorMessage) {
+        Log.e(TAG, "onWebRtcAudioRecordInitError: " + errorMessage);
+        reportError(errorMessage);
+      }
+
+      @Override
+      public void onWebRtcAudioRecordStartError(
+          JavaAudioDeviceModule.AudioRecordStartErrorCode errorCode, String errorMessage) {
+        Log.e(TAG, "onWebRtcAudioRecordStartError: " + errorCode + ". " + errorMessage);
+        reportError(errorMessage);
+      }
+
+      @Override
+      public void onWebRtcAudioRecordError(String errorMessage) {
+        Log.e(TAG, "onWebRtcAudioRecordError: " + errorMessage);
+        reportError(errorMessage);
+      }
+    };
+
+    AudioTrackErrorCallback audioTrackErrorCallback = new AudioTrackErrorCallback() {
+      @Override
+      public void onWebRtcAudioTrackInitError(String errorMessage) {
+        Log.e(TAG, "onWebRtcAudioTrackInitError: " + errorMessage);
+        reportError(errorMessage);
+      }
+
+      @Override
+      public void onWebRtcAudioTrackStartError(
+          JavaAudioDeviceModule.AudioTrackStartErrorCode errorCode, String errorMessage) {
+        Log.e(TAG, "onWebRtcAudioTrackStartError: " + errorCode + ". " + errorMessage);
+        reportError(errorMessage);
+      }
+
+      @Override
+      public void onWebRtcAudioTrackError(String errorMessage) {
+        Log.e(TAG, "onWebRtcAudioTrackError: " + errorMessage);
+        reportError(errorMessage);
+      }
+    };
+
+    // Set audio record state callbacks.
+    AudioRecordStateCallback audioRecordStateCallback = new AudioRecordStateCallback() {
+      @Override
+      public void onWebRtcAudioRecordStart() {
+        Log.i(TAG, "Audio recording starts");
+      }
+
+      @Override
+      public void onWebRtcAudioRecordStop() {
+        Log.i(TAG, "Audio recording stops");
+      }
+    };
+
+    // Set audio track state callbacks.
+    AudioTrackStateCallback audioTrackStateCallback = new AudioTrackStateCallback() {
+      @Override
+      public void onWebRtcAudioTrackStart() {
+        Log.i(TAG, "Audio playout starts");
+      }
+
+      @Override
+      public void onWebRtcAudioTrackStop() {
+        Log.i(TAG, "Audio playout stops");
+      }
+    };
+
+    return JavaAudioDeviceModule.builder(appContext)
+        .setSamplesReadyCallback(saveRecordedAudioToFile)
+        .setUseHardwareAcousticEchoCanceler(!peerConnectionParameters.disableBuiltInAEC)
+        .setUseHardwareNoiseSuppressor(!peerConnectionParameters.disableBuiltInNS)
+        .setAudioRecordErrorCallback(audioRecordErrorCallback)
+        .setAudioTrackErrorCallback(audioTrackErrorCallback)
+        .setAudioRecordStateCallback(audioRecordStateCallback)
+        .setAudioTrackStateCallback(audioTrackStateCallback)
+        .createAudioDeviceModule();
+  }
+
+  private void createMediaConstraintsInternal() {
+    // Create video constraints if video call is enabled.
+    if (isVideoCallEnabled()) {
+      videoWidth = peerConnectionParameters.videoWidth;
+      videoHeight = peerConnectionParameters.videoHeight;
+      videoFps = peerConnectionParameters.videoFps;
+
+      // If video resolution is not specified, default to HD.
+      if (videoWidth == 0 || videoHeight == 0) {
+        videoWidth = HD_VIDEO_WIDTH;
+        videoHeight = HD_VIDEO_HEIGHT;
+      }
+
+      // If fps is not specified, default to 30.
+      if (videoFps == 0) {
+        videoFps = 30;
+      }
+      Logging.d(TAG, "Capturing format: " + videoWidth + "x" + videoHeight + "@" + videoFps);
+    }
+
+    // Create audio constraints.
+    audioConstraints = new MediaConstraints();
+    // added for audio performance measurements
+    if (peerConnectionParameters.noAudioProcessing) {
+      Log.d(TAG, "Disabling audio processing");
+      audioConstraints.mandatory.add(
+          new MediaConstraints.KeyValuePair(AUDIO_ECHO_CANCELLATION_CONSTRAINT, "false"));
+      audioConstraints.mandatory.add(
+          new MediaConstraints.KeyValuePair(AUDIO_AUTO_GAIN_CONTROL_CONSTRAINT, "false"));
+      audioConstraints.mandatory.add(
+          new MediaConstraints.KeyValuePair(AUDIO_HIGH_PASS_FILTER_CONSTRAINT, "false"));
+      audioConstraints.mandatory.add(
+          new MediaConstraints.KeyValuePair(AUDIO_NOISE_SUPPRESSION_CONSTRAINT, "false"));
+    }
+    // Create SDP constraints.
+    sdpMediaConstraints = new MediaConstraints();
+    sdpMediaConstraints.mandatory.add(
+        new MediaConstraints.KeyValuePair("OfferToReceiveAudio", "true"));
+    sdpMediaConstraints.mandatory.add(new MediaConstraints.KeyValuePair(
+        "OfferToReceiveVideo", Boolean.toString(isVideoCallEnabled())));
+  }
+
+  private void createPeerConnectionInternal() {
+    if (factory == null || isError) {
+      Log.e(TAG, "Peerconnection factory is not created");
+      return;
+    }
+    Log.d(TAG, "Create peer connection.");
+
+    queuedRemoteCandidates = new ArrayList<>();
+
+    PeerConnection.RTCConfiguration rtcConfig =
+        new PeerConnection.RTCConfiguration(signalingParameters.iceServers);
+    // TCP candidates are only useful when connecting to a server that supports
+    // ICE-TCP.
+    rtcConfig.tcpCandidatePolicy = PeerConnection.TcpCandidatePolicy.DISABLED;
+    rtcConfig.bundlePolicy = PeerConnection.BundlePolicy.MAXBUNDLE;
+    rtcConfig.rtcpMuxPolicy = PeerConnection.RtcpMuxPolicy.REQUIRE;
+    rtcConfig.continualGatheringPolicy = PeerConnection.ContinualGatheringPolicy.GATHER_CONTINUALLY;
+    // Use ECDSA encryption.
+    rtcConfig.keyType = PeerConnection.KeyType.ECDSA;
+    rtcConfig.sdpSemantics = PeerConnection.SdpSemantics.UNIFIED_PLAN;
+
+    peerConnection = factory.createPeerConnection(rtcConfig, pcObserver);
+
+    if (dataChannelEnabled) {
+      DataChannel.Init init = new DataChannel.Init();
+      init.ordered = peerConnectionParameters.dataChannelParameters.ordered;
+      init.negotiated = peerConnectionParameters.dataChannelParameters.negotiated;
+      init.maxRetransmits = peerConnectionParameters.dataChannelParameters.maxRetransmits;
+      init.maxRetransmitTimeMs = peerConnectionParameters.dataChannelParameters.maxRetransmitTimeMs;
+      init.id = peerConnectionParameters.dataChannelParameters.id;
+      init.protocol = peerConnectionParameters.dataChannelParameters.protocol;
+      dataChannel = peerConnection.createDataChannel("ApprtcDemo data", init);
+    }
+    isInitiator = false;
+
+    // Set INFO libjingle logging.
+    // NOTE: this _must_ happen while `factory` is alive!
+    Logging.enableLogToDebugOutput(Logging.Severity.LS_INFO);
+
+    List<String> mediaStreamLabels = Collections.singletonList("ARDAMS");
+    if (isVideoCallEnabled()) {
+      peerConnection.addTrack(createVideoTrack(videoCapturer), mediaStreamLabels);
+      // We can add the renderers right away because we don't need to wait for an
+      // answer to get the remote track.
+      remoteVideoTrack = getRemoteVideoTrack();
+      remoteVideoTrack.setEnabled(renderVideo);
+      for (VideoSink remoteSink : remoteSinks) {
+        remoteVideoTrack.addSink(remoteSink);
+      }
+    }
+    peerConnection.addTrack(createAudioTrack(), mediaStreamLabels);
+    if (isVideoCallEnabled()) {
+      findVideoSender();
+    }
+
+    if (peerConnectionParameters.aecDump) {
+      try {
+        ParcelFileDescriptor aecDumpFileDescriptor =
+            ParcelFileDescriptor.open(new File(Environment.getExternalStorageDirectory().getPath()
+                                          + File.separator + "Download/audio.aecdump"),
+                ParcelFileDescriptor.MODE_READ_WRITE | ParcelFileDescriptor.MODE_CREATE
+                    | ParcelFileDescriptor.MODE_TRUNCATE);
+        factory.startAecDump(aecDumpFileDescriptor.detachFd(), -1);
+      } catch (IOException e) {
+        Log.e(TAG, "Can not open aecdump file", e);
+      }
+    }
+
+    if (saveRecordedAudioToFile != null) {
+      if (saveRecordedAudioToFile.start()) {
+        Log.d(TAG, "Recording input audio to file is activated");
+      }
+    }
+    Log.d(TAG, "Peer connection created.");
+  }
+
+  private File createRtcEventLogOutputFile() {
+    DateFormat dateFormat = new SimpleDateFormat("yyyyMMdd_hhmm_ss", Locale.getDefault());
+    Date date = new Date();
+    final String outputFileName = "event_log_" + dateFormat.format(date) + ".log";
+    return new File(
+        appContext.getDir(RTCEVENTLOG_OUTPUT_DIR_NAME, Context.MODE_PRIVATE), outputFileName);
+  }
+
+  private void maybeCreateAndStartRtcEventLog() {
+    if (appContext == null || peerConnection == null) {
+      return;
+    }
+    if (!peerConnectionParameters.enableRtcEventLog) {
+      Log.d(TAG, "RtcEventLog is disabled.");
+      return;
+    }
+    rtcEventLog = new RtcEventLog(peerConnection);
+    rtcEventLog.start(createRtcEventLogOutputFile());
+  }
+
+  private void closeInternal() {
+    if (factory != null && peerConnectionParameters.aecDump) {
+      factory.stopAecDump();
+    }
+    Log.d(TAG, "Closing peer connection.");
+    statsTimer.cancel();
+    if (dataChannel != null) {
+      dataChannel.dispose();
+      dataChannel = null;
+    }
+    if (rtcEventLog != null) {
+      // RtcEventLog should stop before the peer connection is disposed.
+      rtcEventLog.stop();
+      rtcEventLog = null;
+    }
+    if (peerConnection != null) {
+      peerConnection.dispose();
+      peerConnection = null;
+    }
+    Log.d(TAG, "Closing audio source.");
+    if (audioSource != null) {
+      audioSource.dispose();
+      audioSource = null;
+    }
+    Log.d(TAG, "Stopping capture.");
+    if (videoCapturer != null) {
+      try {
+        videoCapturer.stopCapture();
+      } catch (InterruptedException e) {
+        throw new RuntimeException(e);
+      }
+      videoCapturerStopped = true;
+      videoCapturer.dispose();
+      videoCapturer = null;
+    }
+    Log.d(TAG, "Closing video source.");
+    if (videoSource != null) {
+      videoSource.dispose();
+      videoSource = null;
+    }
+    if (surfaceTextureHelper != null) {
+      surfaceTextureHelper.dispose();
+      surfaceTextureHelper = null;
+    }
+    if (saveRecordedAudioToFile != null) {
+      Log.d(TAG, "Closing audio file for recorded input audio.");
+      saveRecordedAudioToFile.stop();
+      saveRecordedAudioToFile = null;
+    }
+    localRender = null;
+    remoteSinks = null;
+    Log.d(TAG, "Closing peer connection factory.");
+    if (factory != null) {
+      factory.dispose();
+      factory = null;
+    }
+    rootEglBase.release();
+    Log.d(TAG, "Closing peer connection done.");
+    events.onPeerConnectionClosed();
+    PeerConnectionFactory.stopInternalTracingCapture();
+    PeerConnectionFactory.shutdownInternalTracer();
+  }
+
+  public boolean isHDVideo() {
+    return isVideoCallEnabled() && videoWidth * videoHeight >= 1280 * 720;
+  }
+
+  private void getStats() {
+    if (peerConnection == null || isError) {
+      return;
+    }
+    peerConnection.getStats(new RTCStatsCollectorCallback() {
+      @Override
+      public void onStatsDelivered(RTCStatsReport report) {
+        events.onPeerConnectionStatsReady(report);
+      }
+    });
+  }
+
+  public void enableStatsEvents(boolean enable, int periodMs) {
+    if (enable) {
+      try {
+        statsTimer.schedule(new TimerTask() {
+          @Override
+          public void run() {
+            executor.execute(() -> getStats());
+          }
+        }, 0, periodMs);
+      } catch (Exception e) {
+        Log.e(TAG, "Can not schedule statistics timer", e);
+      }
+    } else {
+      statsTimer.cancel();
+    }
+  }
+
+  public void setAudioEnabled(final boolean enable) {
+    executor.execute(() -> {
+      enableAudio = enable;
+      if (localAudioTrack != null) {
+        localAudioTrack.setEnabled(enableAudio);
+      }
+    });
+  }
+
+  public void setVideoEnabled(final boolean enable) {
+    executor.execute(() -> {
+      renderVideo = enable;
+      if (localVideoTrack != null) {
+        localVideoTrack.setEnabled(renderVideo);
+      }
+      if (remoteVideoTrack != null) {
+        remoteVideoTrack.setEnabled(renderVideo);
+      }
+    });
+  }
+
+  public void createOffer() {
+    executor.execute(() -> {
+      if (peerConnection != null && !isError) {
+        Log.d(TAG, "PC Create OFFER");
+        isInitiator = true;
+        peerConnection.createOffer(sdpObserver, sdpMediaConstraints);
+      }
+    });
+  }
+
+  public void createAnswer() {
+    executor.execute(() -> {
+      if (peerConnection != null && !isError) {
+        Log.d(TAG, "PC create ANSWER");
+        isInitiator = false;
+        peerConnection.createAnswer(sdpObserver, sdpMediaConstraints);
+      }
+    });
+  }
+
+  public void addRemoteIceCandidate(final IceCandidate candidate) {
+    executor.execute(() -> {
+      if (peerConnection != null && !isError) {
+        if (queuedRemoteCandidates != null) {
+          queuedRemoteCandidates.add(candidate);
+        } else {
+          peerConnection.addIceCandidate(candidate, new AddIceObserver() {
+            @Override
+            public void onAddSuccess() {
+              Log.d(TAG, "Candidate " + candidate + " successfully added.");
+            }
+            @Override
+            public void onAddFailure(String error) {
+              Log.d(TAG, "Candidate " + candidate + " addition failed: " + error);
+            }
+          });
+        }
+      }
+    });
+  }
+
+  public void removeRemoteIceCandidates(final IceCandidate[] candidates) {
+    executor.execute(() -> {
+      if (peerConnection == null || isError) {
+        return;
+      }
+      // Drain the queued remote candidates if there is any so that
+      // they are processed in the proper order.
+      drainCandidates();
+      peerConnection.removeIceCandidates(candidates);
+    });
+  }
+
+  public void setRemoteDescription(final SessionDescription desc) {
+    executor.execute(() -> {
+      if (peerConnection == null || isError) {
+        return;
+      }
+      String sdp = desc.description;
+      if (preferIsac) {
+        sdp = preferCodec(sdp, AUDIO_CODEC_ISAC, true);
+      }
+      if (isVideoCallEnabled()) {
+        sdp = preferCodec(sdp, getSdpVideoCodecName(peerConnectionParameters), false);
+      }
+      if (peerConnectionParameters.audioStartBitrate > 0) {
+        sdp = setStartBitrate(
+            AUDIO_CODEC_OPUS, false, sdp, peerConnectionParameters.audioStartBitrate);
+      }
+      Log.d(TAG, "Set remote SDP.");
+      SessionDescription sdpRemote = new SessionDescription(desc.type, sdp);
+      peerConnection.setRemoteDescription(sdpObserver, sdpRemote);
+    });
+  }
+
+  public void stopVideoSource() {
+    executor.execute(() -> {
+      if (videoCapturer != null && !videoCapturerStopped) {
+        Log.d(TAG, "Stop video source.");
+        try {
+          videoCapturer.stopCapture();
+        } catch (InterruptedException e) {
+        }
+        videoCapturerStopped = true;
+      }
+    });
+  }
+
+  public void startVideoSource() {
+    executor.execute(() -> {
+      if (videoCapturer != null && videoCapturerStopped) {
+        Log.d(TAG, "Restart video source.");
+        videoCapturer.startCapture(videoWidth, videoHeight, videoFps);
+        videoCapturerStopped = false;
+      }
+    });
+  }
+
+  public void setVideoMaxBitrate(@Nullable final Integer maxBitrateKbps) {
+    executor.execute(() -> {
+      if (peerConnection == null || localVideoSender == null || isError) {
+        return;
+      }
+      Log.d(TAG, "Requested max video bitrate: " + maxBitrateKbps);
+      if (localVideoSender == null) {
+        Log.w(TAG, "Sender is not ready.");
+        return;
+      }
+
+      RtpParameters parameters = localVideoSender.getParameters();
+      if (parameters.encodings.size() == 0) {
+        Log.w(TAG, "RtpParameters are not ready.");
+        return;
+      }
+
+      for (RtpParameters.Encoding encoding : parameters.encodings) {
+        // Null value means no limit.
+        encoding.maxBitrateBps = maxBitrateKbps == null ? null : maxBitrateKbps * BPS_IN_KBPS;
+      }
+      if (!localVideoSender.setParameters(parameters)) {
+        Log.e(TAG, "RtpSender.setParameters failed.");
+      }
+      Log.d(TAG, "Configured max video bitrate to: " + maxBitrateKbps);
+    });
+  }
+
+  private void reportError(final String errorMessage) {
+    Log.e(TAG, "Peerconnection error: " + errorMessage);
+    executor.execute(() -> {
+      if (!isError) {
+        events.onPeerConnectionError(errorMessage);
+        isError = true;
+      }
+    });
+  }
+
+  @Nullable
+  private AudioTrack createAudioTrack() {
+    audioSource = factory.createAudioSource(audioConstraints);
+    localAudioTrack = factory.createAudioTrack(AUDIO_TRACK_ID, audioSource);
+    localAudioTrack.setEnabled(enableAudio);
+    return localAudioTrack;
+  }
+
+  @Nullable
+  private VideoTrack createVideoTrack(VideoCapturer capturer) {
+    surfaceTextureHelper =
+        SurfaceTextureHelper.create("CaptureThread", rootEglBase.getEglBaseContext());
+    videoSource = factory.createVideoSource(capturer.isScreencast());
+    capturer.initialize(surfaceTextureHelper, appContext, videoSource.getCapturerObserver());
+    capturer.startCapture(videoWidth, videoHeight, videoFps);
+
+    localVideoTrack = factory.createVideoTrack(VIDEO_TRACK_ID, videoSource);
+    localVideoTrack.setEnabled(renderVideo);
+    localVideoTrack.addSink(localRender);
+    return localVideoTrack;
+  }
+
+  private void findVideoSender() {
+    for (RtpSender sender : peerConnection.getSenders()) {
+      if (sender.track() != null) {
+        String trackType = sender.track().kind();
+        if (trackType.equals(VIDEO_TRACK_TYPE)) {
+          Log.d(TAG, "Found video sender.");
+          localVideoSender = sender;
+        }
+      }
+    }
+  }
+
+  // Returns the remote VideoTrack, assuming there is only one.
+  private @Nullable VideoTrack getRemoteVideoTrack() {
+    for (RtpTransceiver transceiver : peerConnection.getTransceivers()) {
+      MediaStreamTrack track = transceiver.getReceiver().track();
+      if (track instanceof VideoTrack) {
+        return (VideoTrack) track;
+      }
+    }
+    return null;
+  }
+
+  private static String getSdpVideoCodecName(PeerConnectionParameters parameters) {
+    switch (parameters.videoCodec) {
+      case VIDEO_CODEC_VP8:
+        return VIDEO_CODEC_VP8;
+      case VIDEO_CODEC_VP9:
+        return VIDEO_CODEC_VP9;
+      case VIDEO_CODEC_AV1:
+        return VIDEO_CODEC_AV1;
+      case VIDEO_CODEC_H264_HIGH:
+      case VIDEO_CODEC_H264_BASELINE:
+        return VIDEO_CODEC_H264;
+      default:
+        return VIDEO_CODEC_VP8;
+    }
+  }
+
+  private static String getFieldTrials(PeerConnectionParameters peerConnectionParameters) {
+    String fieldTrials = "";
+    if (peerConnectionParameters.videoFlexfecEnabled) {
+      fieldTrials += VIDEO_FLEXFEC_FIELDTRIAL;
+      Log.d(TAG, "Enable FlexFEC field trial.");
+    }
+    fieldTrials += VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL;
+    if (peerConnectionParameters.disableWebRtcAGCAndHPF) {
+      fieldTrials += DISABLE_WEBRTC_AGC_FIELDTRIAL;
+      Log.d(TAG, "Disable WebRTC AGC field trial.");
+    }
+    return fieldTrials;
+  }
+
+  @SuppressWarnings("StringSplitter")
+  private static String setStartBitrate(
+      String codec, boolean isVideoCodec, String sdp, int bitrateKbps) {
+    String[] lines = sdp.split("\r\n");
+    int rtpmapLineIndex = -1;
+    boolean sdpFormatUpdated = false;
+    String codecRtpMap = null;
+    // Search for codec rtpmap in format
+    // a=rtpmap:<payload type> <encoding name>/<clock rate> [/<encoding parameters>]
+    String regex = "^a=rtpmap:(\\d+) " + codec + "(/\\d+)+[\r]?$";
+    Pattern codecPattern = Pattern.compile(regex);
+    for (int i = 0; i < lines.length; i++) {
+      Matcher codecMatcher = codecPattern.matcher(lines[i]);
+      if (codecMatcher.matches()) {
+        codecRtpMap = codecMatcher.group(1);
+        rtpmapLineIndex = i;
+        break;
+      }
+    }
+    if (codecRtpMap == null) {
+      Log.w(TAG, "No rtpmap for " + codec + " codec");
+      return sdp;
+    }
+    Log.d(TAG, "Found " + codec + " rtpmap " + codecRtpMap + " at " + lines[rtpmapLineIndex]);
+
+    // Check if a=fmtp string already exist in remote SDP for this codec and
+    // update it with new bitrate parameter.
+    regex = "^a=fmtp:" + codecRtpMap + " \\w+=\\d+.*[\r]?$";
+    codecPattern = Pattern.compile(regex);
+    for (int i = 0; i < lines.length; i++) {
+      Matcher codecMatcher = codecPattern.matcher(lines[i]);
+      if (codecMatcher.matches()) {
+        Log.d(TAG, "Found " + codec + " " + lines[i]);
+        if (isVideoCodec) {
+          lines[i] += "; " + VIDEO_CODEC_PARAM_START_BITRATE + "=" + bitrateKbps;
+        } else {
+          lines[i] += "; " + AUDIO_CODEC_PARAM_BITRATE + "=" + (bitrateKbps * 1000);
+        }
+        Log.d(TAG, "Update remote SDP line: " + lines[i]);
+        sdpFormatUpdated = true;
+        break;
+      }
+    }
+
+    StringBuilder newSdpDescription = new StringBuilder();
+    for (int i = 0; i < lines.length; i++) {
+      newSdpDescription.append(lines[i]).append("\r\n");
+      // Append new a=fmtp line if no such line exist for a codec.
+      if (!sdpFormatUpdated && i == rtpmapLineIndex) {
+        String bitrateSet;
+        if (isVideoCodec) {
+          bitrateSet =
+              "a=fmtp:" + codecRtpMap + " " + VIDEO_CODEC_PARAM_START_BITRATE + "=" + bitrateKbps;
+        } else {
+          bitrateSet = "a=fmtp:" + codecRtpMap + " " + AUDIO_CODEC_PARAM_BITRATE + "="
+              + (bitrateKbps * 1000);
+        }
+        Log.d(TAG, "Add remote SDP line: " + bitrateSet);
+        newSdpDescription.append(bitrateSet).append("\r\n");
+      }
+    }
+    return newSdpDescription.toString();
+  }
+
+  /** Returns the line number containing "m=audio|video", or -1 if no such line exists. */
+  private static int findMediaDescriptionLine(boolean isAudio, String[] sdpLines) {
+    final String mediaDescription = isAudio ? "m=audio " : "m=video ";
+    for (int i = 0; i < sdpLines.length; ++i) {
+      if (sdpLines[i].startsWith(mediaDescription)) {
+        return i;
+      }
+    }
+    return -1;
+  }
+
+  private static String joinString(
+      Iterable<? extends CharSequence> s, String delimiter, boolean delimiterAtEnd) {
+    Iterator<? extends CharSequence> iter = s.iterator();
+    if (!iter.hasNext()) {
+      return "";
+    }
+    StringBuilder buffer = new StringBuilder(iter.next());
+    while (iter.hasNext()) {
+      buffer.append(delimiter).append(iter.next());
+    }
+    if (delimiterAtEnd) {
+      buffer.append(delimiter);
+    }
+    return buffer.toString();
+  }
+
+  private static @Nullable String movePayloadTypesToFront(
+      List<String> preferredPayloadTypes, String mLine) {
+    // The format of the media description line should be: m=<media> <port> <proto> <fmt> ...
+    final List<String> origLineParts = Arrays.asList(mLine.split(" "));
+    if (origLineParts.size() <= 3) {
+      Log.e(TAG, "Wrong SDP media description format: " + mLine);
+      return null;
+    }
+    final List<String> header = origLineParts.subList(0, 3);
+    final List<String> unpreferredPayloadTypes =
+        new ArrayList<>(origLineParts.subList(3, origLineParts.size()));
+    unpreferredPayloadTypes.removeAll(preferredPayloadTypes);
+    // Reconstruct the line with `preferredPayloadTypes` moved to the beginning of the payload
+    // types.
+    final List<String> newLineParts = new ArrayList<>();
+    newLineParts.addAll(header);
+    newLineParts.addAll(preferredPayloadTypes);
+    newLineParts.addAll(unpreferredPayloadTypes);
+    return joinString(newLineParts, " ", false /* delimiterAtEnd */);
+  }
+
+  private static String preferCodec(String sdp, String codec, boolean isAudio) {
+    final String[] lines = sdp.split("\r\n");
+    final int mLineIndex = findMediaDescriptionLine(isAudio, lines);
+    if (mLineIndex == -1) {
+      Log.w(TAG, "No mediaDescription line, so can't prefer " + codec);
+      return sdp;
+    }
+    // A list with all the payload types with name `codec`. The payload types are integers in the
+    // range 96-127, but they are stored as strings here.
+    final List<String> codecPayloadTypes = new ArrayList<>();
+    // a=rtpmap:<payload type> <encoding name>/<clock rate> [/<encoding parameters>]
+    final Pattern codecPattern = Pattern.compile("^a=rtpmap:(\\d+) " + codec + "(/\\d+)+[\r]?$");
+    for (String line : lines) {
+      Matcher codecMatcher = codecPattern.matcher(line);
+      if (codecMatcher.matches()) {
+        codecPayloadTypes.add(codecMatcher.group(1));
+      }
+    }
+    if (codecPayloadTypes.isEmpty()) {
+      Log.w(TAG, "No payload types with name " + codec);
+      return sdp;
+    }
+
+    final String newMLine = movePayloadTypesToFront(codecPayloadTypes, lines[mLineIndex]);
+    if (newMLine == null) {
+      return sdp;
+    }
+    Log.d(TAG, "Change media description from: " + lines[mLineIndex] + " to " + newMLine);
+    lines[mLineIndex] = newMLine;
+    return joinString(Arrays.asList(lines), "\r\n", true /* delimiterAtEnd */);
+  }
+
+  private void drainCandidates() {
+    if (queuedRemoteCandidates != null) {
+      Log.d(TAG, "Add " + queuedRemoteCandidates.size() + " remote candidates");
+      for (IceCandidate candidate : queuedRemoteCandidates) {
+        peerConnection.addIceCandidate(candidate, new AddIceObserver() {
+          @Override
+          public void onAddSuccess() {
+            Log.d(TAG, "Candidate " + candidate + " successfully added.");
+          }
+          @Override
+          public void onAddFailure(String error) {
+            Log.d(TAG, "Candidate " + candidate + " addition failed: " + error);
+          }
+        });
+      }
+      queuedRemoteCandidates = null;
+    }
+  }
+
+  private void switchCameraInternal() {
+    if (videoCapturer instanceof CameraVideoCapturer) {
+      if (!isVideoCallEnabled() || isError) {
+        Log.e(TAG,
+            "Failed to switch camera. Video: " + isVideoCallEnabled() + ". Error : " + isError);
+        return; // No video is sent or only one camera is available or error happened.
+      }
+      Log.d(TAG, "Switch camera");
+      CameraVideoCapturer cameraVideoCapturer = (CameraVideoCapturer) videoCapturer;
+      cameraVideoCapturer.switchCamera(null);
+    } else {
+      Log.d(TAG, "Will not switch camera, video caputurer is not a camera");
+    }
+  }
+
+  public void switchCamera() {
+    executor.execute(this ::switchCameraInternal);
+  }
+
+  public void changeCaptureFormat(final int width, final int height, final int framerate) {
+    executor.execute(() -> changeCaptureFormatInternal(width, height, framerate));
+  }
+
+  private void changeCaptureFormatInternal(int width, int height, int framerate) {
+    if (!isVideoCallEnabled() || isError || videoCapturer == null) {
+      Log.e(TAG,
+          "Failed to change capture format. Video: " + isVideoCallEnabled()
+              + ". Error : " + isError);
+      return;
+    }
+    Log.d(TAG, "changeCaptureFormat: " + width + "x" + height + "@" + framerate);
+    videoSource.adaptOutputFormat(width, height, framerate);
+  }
+
+  // Implementation detail: observe ICE & stream changes and react accordingly.
+  private class PCObserver implements PeerConnection.Observer {
+    @Override
+    public void onIceCandidate(final IceCandidate candidate) {
+      executor.execute(() -> events.onIceCandidate(candidate));
+    }
+
+    @Override
+    public void onIceCandidateError(final IceCandidateErrorEvent event) {
+      Log.d(TAG,
+          "IceCandidateError address: " + event.address + ", port: " + event.port + ", url: "
+              + event.url + ", errorCode: " + event.errorCode + ", errorText: " + event.errorText);
+    }
+
+    @Override
+    public void onIceCandidatesRemoved(final IceCandidate[] candidates) {
+      executor.execute(() -> events.onIceCandidatesRemoved(candidates));
+    }
+
+    @Override
+    public void onSignalingChange(PeerConnection.SignalingState newState) {
+      Log.d(TAG, "SignalingState: " + newState);
+    }
+
+    @Override
+    public void onIceConnectionChange(final PeerConnection.IceConnectionState newState) {
+      executor.execute(() -> {
+        Log.d(TAG, "IceConnectionState: " + newState);
+        if (newState == IceConnectionState.CONNECTED) {
+          events.onIceConnected();
+        } else if (newState == IceConnectionState.DISCONNECTED) {
+          events.onIceDisconnected();
+        } else if (newState == IceConnectionState.FAILED) {
+          reportError("ICE connection failed.");
+        }
+      });
+    }
+
+    @Override
+    public void onConnectionChange(final PeerConnection.PeerConnectionState newState) {
+      executor.execute(() -> {
+        Log.d(TAG, "PeerConnectionState: " + newState);
+        if (newState == PeerConnectionState.CONNECTED) {
+          events.onConnected();
+        } else if (newState == PeerConnectionState.DISCONNECTED) {
+          events.onDisconnected();
+        } else if (newState == PeerConnectionState.FAILED) {
+          reportError("DTLS connection failed.");
+        }
+      });
+    }
+
+    @Override
+    public void onIceGatheringChange(PeerConnection.IceGatheringState newState) {
+      Log.d(TAG, "IceGatheringState: " + newState);
+    }
+
+    @Override
+    public void onIceConnectionReceivingChange(boolean receiving) {
+      Log.d(TAG, "IceConnectionReceiving changed to " + receiving);
+    }
+
+    @Override
+    public void onSelectedCandidatePairChanged(CandidatePairChangeEvent event) {
+      Log.d(TAG, "Selected candidate pair changed because: " + event);
+    }
+
+    @Override
+    public void onAddStream(final MediaStream stream) {}
+
+    @Override
+    public void onRemoveStream(final MediaStream stream) {}
+
+    @Override
+    public void onDataChannel(final DataChannel dc) {
+      Log.d(TAG, "New Data channel " + dc.label());
+
+      if (!dataChannelEnabled)
+        return;
+
+      dc.registerObserver(new DataChannel.Observer() {
+        @Override
+        public void onBufferedAmountChange(long previousAmount) {
+          Log.d(TAG, "Data channel buffered amount changed: " + dc.label() + ": " + dc.state());
+        }
+
+        @Override
+        public void onStateChange() {
+          Log.d(TAG, "Data channel state changed: " + dc.label() + ": " + dc.state());
+        }
+
+        @Override
+        public void onMessage(final DataChannel.Buffer buffer) {
+          if (buffer.binary) {
+            Log.d(TAG, "Received binary msg over " + dc);
+            return;
+          }
+          ByteBuffer data = buffer.data;
+          final byte[] bytes = new byte[data.capacity()];
+          data.get(bytes);
+          String strData = new String(bytes, Charset.forName("UTF-8"));
+          Log.d(TAG, "Got msg: " + strData + " over " + dc);
+        }
+      });
+    }
+
+    @Override
+    public void onRenegotiationNeeded() {
+      // No need to do anything; AppRTC follows a pre-agreed-upon
+      // signaling/negotiation protocol.
+    }
+
+    @Override
+    public void onAddTrack(final RtpReceiver receiver, final MediaStream[] mediaStreams) {}
+
+    @Override
+    public void onRemoveTrack(final RtpReceiver receiver) {}
+  }
+
+  // Implementation detail: handle offer creation/signaling and answer setting,
+  // as well as adding remote ICE candidates once the answer SDP is set.
+  private class SDPObserver implements SdpObserver {
+    @Override
+    public void onCreateSuccess(final SessionDescription desc) {
+      if (localDescription != null) {
+        reportError("Multiple SDP create.");
+        return;
+      }
+      String sdp = desc.description;
+      if (preferIsac) {
+        sdp = preferCodec(sdp, AUDIO_CODEC_ISAC, true);
+      }
+      if (isVideoCallEnabled()) {
+        sdp = preferCodec(sdp, getSdpVideoCodecName(peerConnectionParameters), false);
+      }
+      final SessionDescription newDesc = new SessionDescription(desc.type, sdp);
+      localDescription = newDesc;
+      executor.execute(() -> {
+        if (peerConnection != null && !isError) {
+          Log.d(TAG, "Set local SDP from " + desc.type);
+          peerConnection.setLocalDescription(sdpObserver, newDesc);
+        }
+      });
+    }
+
+    @Override
+    public void onSetSuccess() {
+      executor.execute(() -> {
+        if (peerConnection == null || isError) {
+          return;
+        }
+        if (isInitiator) {
+          // For offering peer connection we first create offer and set
+          // local SDP, then after receiving answer set remote SDP.
+          if (peerConnection.getRemoteDescription() == null) {
+            // We've just set our local SDP so time to send it.
+            Log.d(TAG, "Local SDP set succesfully");
+            events.onLocalDescription(localDescription);
+          } else {
+            // We've just set remote description, so drain remote
+            // and send local ICE candidates.
+            Log.d(TAG, "Remote SDP set succesfully");
+            drainCandidates();
+          }
+        } else {
+          // For answering peer connection we set remote SDP and then
+          // create answer and set local SDP.
+          if (peerConnection.getLocalDescription() != null) {
+            // We've just set our local SDP so time to send it, drain
+            // remote and send local ICE candidates.
+            Log.d(TAG, "Local SDP set succesfully");
+            events.onLocalDescription(localDescription);
+            drainCandidates();
+          } else {
+            // We've just set remote SDP - do nothing for now -
+            // answer will be created soon.
+            Log.d(TAG, "Remote SDP set succesfully");
+          }
+        }
+      });
+    }
+
+    @Override
+    public void onCreateFailure(final String error) {
+      reportError("createSDP error: " + error);
+    }
+
+    @Override
+    public void onSetFailure(final String error) {
+      reportError("setSDP error: " + error);
+    }
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RecordedAudioToFileController.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RecordedAudioToFileController.java
new file mode 100644
index 0000000000..9787852feb
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RecordedAudioToFileController.java
@@ -0,0 +1,143 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.media.AudioFormat;
+import android.os.Environment;
+import android.util.Log;
+import androidx.annotation.Nullable;
+import java.io.File;
+import java.io.FileNotFoundException;
+import java.io.FileOutputStream;
+import java.io.IOException;
+import java.io.OutputStream;
+import java.util.concurrent.ExecutorService;
+import org.webrtc.audio.JavaAudioDeviceModule;
+import org.webrtc.audio.JavaAudioDeviceModule.SamplesReadyCallback;
+
+/**
+ * Implements the AudioRecordSamplesReadyCallback interface and writes
+ * recorded raw audio samples to an output file.
+ */
+public class RecordedAudioToFileController implements SamplesReadyCallback {
+  private static final String TAG = "RecordedAudioToFile";
+  private static final long MAX_FILE_SIZE_IN_BYTES = 58348800L;
+
+  private final Object lock = new Object();
+  private final ExecutorService executor;
+  @Nullable private OutputStream rawAudioFileOutputStream;
+  private boolean isRunning;
+  private long fileSizeInBytes;
+
+  public RecordedAudioToFileController(ExecutorService executor) {
+    Log.d(TAG, "ctor");
+    this.executor = executor;
+  }
+
+  /**
+   * Should be called on the same executor thread as the one provided at
+   * construction.
+   */
+  public boolean start() {
+    Log.d(TAG, "start");
+    if (!isExternalStorageWritable()) {
+      Log.e(TAG, "Writing to external media is not possible");
+      return false;
+    }
+    synchronized (lock) {
+      isRunning = true;
+    }
+    return true;
+  }
+
+  /**
+   * Should be called on the same executor thread as the one provided at
+   * construction.
+   */
+  public void stop() {
+    Log.d(TAG, "stop");
+    synchronized (lock) {
+      isRunning = false;
+      if (rawAudioFileOutputStream != null) {
+        try {
+          rawAudioFileOutputStream.close();
+        } catch (IOException e) {
+          Log.e(TAG, "Failed to close file with saved input audio: " + e);
+        }
+        rawAudioFileOutputStream = null;
+      }
+      fileSizeInBytes = 0;
+    }
+  }
+
+  // Checks if external storage is available for read and write.
+  private boolean isExternalStorageWritable() {
+    String state = Environment.getExternalStorageState();
+    if (Environment.MEDIA_MOUNTED.equals(state)) {
+      return true;
+    }
+    return false;
+  }
+
+  // Utilizes audio parameters to create a file name which contains sufficient
+  // information so that the file can be played using an external file player.
+  // Example: /sdcard/recorded_audio_16bits_48000Hz_mono.pcm.
+  private void openRawAudioOutputFile(int sampleRate, int channelCount) {
+    final String fileName = Environment.getExternalStorageDirectory().getPath() + File.separator
+        + "recorded_audio_16bits_" + String.valueOf(sampleRate) + "Hz"
+        + ((channelCount == 1) ? "_mono" : "_stereo") + ".pcm";
+    final File outputFile = new File(fileName);
+    try {
+      rawAudioFileOutputStream = new FileOutputStream(outputFile);
+    } catch (FileNotFoundException e) {
+      Log.e(TAG, "Failed to open audio output file: " + e.getMessage());
+    }
+    Log.d(TAG, "Opened file for recording: " + fileName);
+  }
+
+  // Called when new audio samples are ready.
+  @Override
+  public void onWebRtcAudioRecordSamplesReady(JavaAudioDeviceModule.AudioSamples samples) {
+    // The native audio layer on Android should use 16-bit PCM format.
+    if (samples.getAudioFormat() != AudioFormat.ENCODING_PCM_16BIT) {
+      Log.e(TAG, "Invalid audio format");
+      return;
+    }
+    synchronized (lock) {
+      // Abort early if stop() has been called.
+      if (!isRunning) {
+        return;
+      }
+      // Open a new file for the first callback only since it allows us to add audio parameters to
+      // the file name.
+      if (rawAudioFileOutputStream == null) {
+        openRawAudioOutputFile(samples.getSampleRate(), samples.getChannelCount());
+        fileSizeInBytes = 0;
+      }
+    }
+    // Append the recorded 16-bit audio samples to the open output file.
+    executor.execute(() -> {
+      if (rawAudioFileOutputStream != null) {
+        try {
+          // Set a limit on max file size. 58348800 bytes corresponds to
+          // approximately 10 minutes of recording in mono at 48kHz.
+          if (fileSizeInBytes < MAX_FILE_SIZE_IN_BYTES) {
+            // Writes samples.getData().length bytes to output stream.
+            rawAudioFileOutputStream.write(samples.getData());
+            fileSizeInBytes += samples.getData().length;
+          }
+        } catch (IOException e) {
+          Log.e(TAG, "Failed to write audio to file: " + e.getMessage());
+        }
+      }
+    });
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RoomParametersFetcher.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RoomParametersFetcher.java
new file mode 100644
index 0000000000..6a0f235528
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RoomParametersFetcher.java
@@ -0,0 +1,226 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.util.Log;
+import java.io.IOException;
+import java.io.InputStream;
+import java.net.HttpURLConnection;
+import java.net.URL;
+import java.util.ArrayList;
+import java.util.Scanner;
+import java.util.List;
+import org.appspot.apprtc.AppRTCClient.SignalingParameters;
+import org.appspot.apprtc.util.AsyncHttpURLConnection;
+import org.appspot.apprtc.util.AsyncHttpURLConnection.AsyncHttpEvents;
+import org.json.JSONArray;
+import org.json.JSONException;
+import org.json.JSONObject;
+import org.webrtc.IceCandidate;
+import org.webrtc.PeerConnection;
+import org.webrtc.SessionDescription;
+
+/**
+ * AsyncTask that converts an AppRTC room URL into the set of signaling
+ * parameters to use with that room.
+ */
+public class RoomParametersFetcher {
+  private static final String TAG = "RoomRTCClient";
+  private static final int TURN_HTTP_TIMEOUT_MS = 5000;
+  private final RoomParametersFetcherEvents events;
+  private final String roomUrl;
+  private final String roomMessage;
+
+  /**
+   * Room parameters fetcher callbacks.
+   */
+  public interface RoomParametersFetcherEvents {
+    /**
+     * Callback fired once the room's signaling parameters
+     * SignalingParameters are extracted.
+     */
+    void onSignalingParametersReady(final SignalingParameters params);
+
+    /**
+     * Callback for room parameters extraction error.
+     */
+    void onSignalingParametersError(final String description);
+  }
+
+  public RoomParametersFetcher(
+      String roomUrl, String roomMessage, final RoomParametersFetcherEvents events) {
+    this.roomUrl = roomUrl;
+    this.roomMessage = roomMessage;
+    this.events = events;
+  }
+
+  public void makeRequest() {
+    Log.d(TAG, "Connecting to room: " + roomUrl);
+    AsyncHttpURLConnection httpConnection =
+        new AsyncHttpURLConnection("POST", roomUrl, roomMessage, new AsyncHttpEvents() {
+          @Override
+          public void onHttpError(String errorMessage) {
+            Log.e(TAG, "Room connection error: " + errorMessage);
+            events.onSignalingParametersError(errorMessage);
+          }
+
+          @Override
+          public void onHttpComplete(String response) {
+            roomHttpResponseParse(response);
+          }
+        });
+    httpConnection.send();
+  }
+
+  private void roomHttpResponseParse(String response) {
+    Log.d(TAG, "Room response: " + response);
+    try {
+      List<IceCandidate> iceCandidates = null;
+      SessionDescription offerSdp = null;
+      JSONObject roomJson = new JSONObject(response);
+
+      String result = roomJson.getString("result");
+      if (!result.equals("SUCCESS")) {
+        events.onSignalingParametersError("Room response error: " + result);
+        return;
+      }
+      response = roomJson.getString("params");
+      roomJson = new JSONObject(response);
+      String roomId = roomJson.getString("room_id");
+      String clientId = roomJson.getString("client_id");
+      String wssUrl = roomJson.getString("wss_url");
+      String wssPostUrl = roomJson.getString("wss_post_url");
+      boolean initiator = (roomJson.getBoolean("is_initiator"));
+      if (!initiator) {
+        iceCandidates = new ArrayList<>();
+        String messagesString = roomJson.getString("messages");
+        JSONArray messages = new JSONArray(messagesString);
+        for (int i = 0; i < messages.length(); ++i) {
+          String messageString = messages.getString(i);
+          JSONObject message = new JSONObject(messageString);
+          String messageType = message.getString("type");
+          Log.d(TAG, "GAE->C #" + i + " : " + messageString);
+          if (messageType.equals("offer")) {
+            offerSdp = new SessionDescription(
+                SessionDescription.Type.fromCanonicalForm(messageType), message.getString("sdp"));
+          } else if (messageType.equals("candidate")) {
+            IceCandidate candidate = new IceCandidate(
+                message.getString("id"), message.getInt("label"), message.getString("candidate"));
+            iceCandidates.add(candidate);
+          } else {
+            Log.e(TAG, "Unknown message: " + messageString);
+          }
+        }
+      }
+      Log.d(TAG, "RoomId: " + roomId + ". ClientId: " + clientId);
+      Log.d(TAG, "Initiator: " + initiator);
+      Log.d(TAG, "WSS url: " + wssUrl);
+      Log.d(TAG, "WSS POST url: " + wssPostUrl);
+
+      List<PeerConnection.IceServer> iceServers =
+          iceServersFromPCConfigJSON(roomJson.getString("pc_config"));
+      boolean isTurnPresent = false;
+      for (PeerConnection.IceServer server : iceServers) {
+        Log.d(TAG, "IceServer: " + server);
+        for (String uri : server.urls) {
+          if (uri.startsWith("turn:")) {
+            isTurnPresent = true;
+            break;
+          }
+        }
+      }
+      // Request TURN servers.
+      if (!isTurnPresent && !roomJson.optString("ice_server_url").isEmpty()) {
+        List<PeerConnection.IceServer> turnServers =
+            requestTurnServers(roomJson.getString("ice_server_url"));
+        for (PeerConnection.IceServer turnServer : turnServers) {
+          Log.d(TAG, "TurnServer: " + turnServer);
+          iceServers.add(turnServer);
+        }
+      }
+
+      SignalingParameters params = new SignalingParameters(
+          iceServers, initiator, clientId, wssUrl, wssPostUrl, offerSdp, iceCandidates);
+      events.onSignalingParametersReady(params);
+    } catch (JSONException e) {
+      events.onSignalingParametersError("Room JSON parsing error: " + e.toString());
+    } catch (IOException e) {
+      events.onSignalingParametersError("Room IO error: " + e.toString());
+    }
+  }
+
+  // Requests & returns a TURN ICE Server based on a request URL.  Must be run
+  // off the main thread!
+  @SuppressWarnings("UseNetworkAnnotations")
+  private List<PeerConnection.IceServer> requestTurnServers(String url)
+      throws IOException, JSONException {
+    List<PeerConnection.IceServer> turnServers = new ArrayList<>();
+    Log.d(TAG, "Request TURN from: " + url);
+    HttpURLConnection connection = (HttpURLConnection) new URL(url).openConnection();
+    connection.setDoOutput(true);
+    connection.setRequestProperty("REFERER", "https://appr.tc");
+    connection.setConnectTimeout(TURN_HTTP_TIMEOUT_MS);
+    connection.setReadTimeout(TURN_HTTP_TIMEOUT_MS);
+    int responseCode = connection.getResponseCode();
+    if (responseCode != 200) {
+      throw new IOException("Non-200 response when requesting TURN server from " + url + " : "
+          + connection.getHeaderField(null));
+    }
+    InputStream responseStream = connection.getInputStream();
+    String response = drainStream(responseStream);
+    connection.disconnect();
+    Log.d(TAG, "TURN response: " + response);
+    JSONObject responseJSON = new JSONObject(response);
+    JSONArray iceServers = responseJSON.getJSONArray("iceServers");
+    for (int i = 0; i < iceServers.length(); ++i) {
+      JSONObject server = iceServers.getJSONObject(i);
+      JSONArray turnUrls = server.getJSONArray("urls");
+      String username = server.has("username") ? server.getString("username") : "";
+      String credential = server.has("credential") ? server.getString("credential") : "";
+      for (int j = 0; j < turnUrls.length(); j++) {
+        String turnUrl = turnUrls.getString(j);
+        PeerConnection.IceServer turnServer =
+            PeerConnection.IceServer.builder(turnUrl)
+              .setUsername(username)
+              .setPassword(credential)
+              .createIceServer();
+        turnServers.add(turnServer);
+      }
+    }
+    return turnServers;
+  }
+
+  // Return the list of ICE servers described by a WebRTCPeerConnection
+  // configuration string.
+  private List<PeerConnection.IceServer> iceServersFromPCConfigJSON(String pcConfig)
+      throws JSONException {
+    JSONObject json = new JSONObject(pcConfig);
+    JSONArray servers = json.getJSONArray("iceServers");
+    List<PeerConnection.IceServer> ret = new ArrayList<>();
+    for (int i = 0; i < servers.length(); ++i) {
+      JSONObject server = servers.getJSONObject(i);
+      String url = server.getString("urls");
+      String credential = server.has("credential") ? server.getString("credential") : "";
+        PeerConnection.IceServer turnServer =
+            PeerConnection.IceServer.builder(url)
+              .setPassword(credential)
+              .createIceServer();
+      ret.add(turnServer);
+    }
+    return ret;
+  }
+
+  // Return the contents of an InputStream as a String.
+  private static String drainStream(InputStream in) {
+    Scanner s = new Scanner(in, "UTF-8").useDelimiter("\\A");
+    return s.hasNext() ? s.next() : "";
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RtcEventLog.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RtcEventLog.java
new file mode 100644
index 0000000000..103ad10f0b
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/RtcEventLog.java
@@ -0,0 +1,73 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.os.ParcelFileDescriptor;
+import android.util.Log;
+import java.io.File;
+import java.io.IOException;
+import org.webrtc.PeerConnection;
+
+public class RtcEventLog {
+  private static final String TAG = "RtcEventLog";
+  private static final int OUTPUT_FILE_MAX_BYTES = 10_000_000;
+  private final PeerConnection peerConnection;
+  private RtcEventLogState state = RtcEventLogState.INACTIVE;
+
+  enum RtcEventLogState {
+    INACTIVE,
+    STARTED,
+    STOPPED,
+  }
+
+  public RtcEventLog(PeerConnection peerConnection) {
+    if (peerConnection == null) {
+      throw new NullPointerException("The peer connection is null.");
+    }
+    this.peerConnection = peerConnection;
+  }
+
+  public void start(final File outputFile) {
+    if (state == RtcEventLogState.STARTED) {
+      Log.e(TAG, "RtcEventLog has already started.");
+      return;
+    }
+    final ParcelFileDescriptor fileDescriptor;
+    try {
+      fileDescriptor = ParcelFileDescriptor.open(outputFile,
+          ParcelFileDescriptor.MODE_READ_WRITE | ParcelFileDescriptor.MODE_CREATE
+              | ParcelFileDescriptor.MODE_TRUNCATE);
+    } catch (IOException e) {
+      Log.e(TAG, "Failed to create a new file", e);
+      return;
+    }
+
+    // Passes ownership of the file to WebRTC.
+    boolean success =
+        peerConnection.startRtcEventLog(fileDescriptor.detachFd(), OUTPUT_FILE_MAX_BYTES);
+    if (!success) {
+      Log.e(TAG, "Failed to start RTC event log.");
+      return;
+    }
+    state = RtcEventLogState.STARTED;
+    Log.d(TAG, "RtcEventLog started.");
+  }
+
+  public void stop() {
+    if (state != RtcEventLogState.STARTED) {
+      Log.e(TAG, "RtcEventLog was not started.");
+      return;
+    }
+    peerConnection.stopRtcEventLog();
+    state = RtcEventLogState.STOPPED;
+    Log.d(TAG, "RtcEventLog stopped.");
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/SettingsActivity.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/SettingsActivity.java
new file mode 100644
index 0000000000..e9c6f6b798
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/SettingsActivity.java
@@ -0,0 +1,317 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.app.Activity;
+import android.content.SharedPreferences;
+import android.content.SharedPreferences.OnSharedPreferenceChangeListener;
+import android.os.Bundle;
+import android.preference.ListPreference;
+import android.preference.Preference;
+import org.webrtc.Camera2Enumerator;
+import org.webrtc.audio.JavaAudioDeviceModule;
+
+/**
+ * Settings activity for AppRTC.
+ */
+public class SettingsActivity extends Activity implements OnSharedPreferenceChangeListener {
+  private SettingsFragment settingsFragment;
+  private String keyprefVideoCall;
+  private String keyprefScreencapture;
+  private String keyprefCamera2;
+  private String keyprefResolution;
+  private String keyprefFps;
+  private String keyprefCaptureQualitySlider;
+  private String keyprefMaxVideoBitrateType;
+  private String keyprefMaxVideoBitrateValue;
+  private String keyPrefVideoCodec;
+  private String keyprefHwCodec;
+  private String keyprefCaptureToTexture;
+  private String keyprefFlexfec;
+
+  private String keyprefStartAudioBitrateType;
+  private String keyprefStartAudioBitrateValue;
+  private String keyPrefAudioCodec;
+  private String keyprefNoAudioProcessing;
+  private String keyprefAecDump;
+  private String keyprefEnableSaveInputAudioToFile;
+  private String keyprefOpenSLES;
+  private String keyprefDisableBuiltInAEC;
+  private String keyprefDisableBuiltInAGC;
+  private String keyprefDisableBuiltInNS;
+  private String keyprefDisableWebRtcAGCAndHPF;
+  private String keyprefSpeakerphone;
+
+  private String keyPrefRoomServerUrl;
+  private String keyPrefDisplayHud;
+  private String keyPrefTracing;
+  private String keyprefEnabledRtcEventLog;
+
+  private String keyprefEnableDataChannel;
+  private String keyprefOrdered;
+  private String keyprefMaxRetransmitTimeMs;
+  private String keyprefMaxRetransmits;
+  private String keyprefDataProtocol;
+  private String keyprefNegotiated;
+  private String keyprefDataId;
+
+  @Override
+  protected void onCreate(Bundle savedInstanceState) {
+    super.onCreate(savedInstanceState);
+    keyprefVideoCall = getString(R.string.pref_videocall_key);
+    keyprefScreencapture = getString(R.string.pref_screencapture_key);
+    keyprefCamera2 = getString(R.string.pref_camera2_key);
+    keyprefResolution = getString(R.string.pref_resolution_key);
+    keyprefFps = getString(R.string.pref_fps_key);
+    keyprefCaptureQualitySlider = getString(R.string.pref_capturequalityslider_key);
+    keyprefMaxVideoBitrateType = getString(R.string.pref_maxvideobitrate_key);
+    keyprefMaxVideoBitrateValue = getString(R.string.pref_maxvideobitratevalue_key);
+    keyPrefVideoCodec = getString(R.string.pref_videocodec_key);
+    keyprefHwCodec = getString(R.string.pref_hwcodec_key);
+    keyprefCaptureToTexture = getString(R.string.pref_capturetotexture_key);
+    keyprefFlexfec = getString(R.string.pref_flexfec_key);
+
+    keyprefStartAudioBitrateType = getString(R.string.pref_startaudiobitrate_key);
+    keyprefStartAudioBitrateValue = getString(R.string.pref_startaudiobitratevalue_key);
+    keyPrefAudioCodec = getString(R.string.pref_audiocodec_key);
+    keyprefNoAudioProcessing = getString(R.string.pref_noaudioprocessing_key);
+    keyprefAecDump = getString(R.string.pref_aecdump_key);
+    keyprefEnableSaveInputAudioToFile =
+        getString(R.string.pref_enable_save_input_audio_to_file_key);
+    keyprefOpenSLES = getString(R.string.pref_opensles_key);
+    keyprefDisableBuiltInAEC = getString(R.string.pref_disable_built_in_aec_key);
+    keyprefDisableBuiltInAGC = getString(R.string.pref_disable_built_in_agc_key);
+    keyprefDisableBuiltInNS = getString(R.string.pref_disable_built_in_ns_key);
+    keyprefDisableWebRtcAGCAndHPF = getString(R.string.pref_disable_webrtc_agc_and_hpf_key);
+    keyprefSpeakerphone = getString(R.string.pref_speakerphone_key);
+
+    keyprefEnableDataChannel = getString(R.string.pref_enable_datachannel_key);
+    keyprefOrdered = getString(R.string.pref_ordered_key);
+    keyprefMaxRetransmitTimeMs = getString(R.string.pref_max_retransmit_time_ms_key);
+    keyprefMaxRetransmits = getString(R.string.pref_max_retransmits_key);
+    keyprefDataProtocol = getString(R.string.pref_data_protocol_key);
+    keyprefNegotiated = getString(R.string.pref_negotiated_key);
+    keyprefDataId = getString(R.string.pref_data_id_key);
+
+    keyPrefRoomServerUrl = getString(R.string.pref_room_server_url_key);
+    keyPrefDisplayHud = getString(R.string.pref_displayhud_key);
+    keyPrefTracing = getString(R.string.pref_tracing_key);
+    keyprefEnabledRtcEventLog = getString(R.string.pref_enable_rtceventlog_key);
+
+    // Display the fragment as the main content.
+    settingsFragment = new SettingsFragment();
+    getFragmentManager()
+        .beginTransaction()
+        .replace(android.R.id.content, settingsFragment)
+        .commit();
+  }
+
+  @Override
+  protected void onResume() {
+    super.onResume();
+    // Set summary to be the user-description for the selected value
+    SharedPreferences sharedPreferences =
+        settingsFragment.getPreferenceScreen().getSharedPreferences();
+    sharedPreferences.registerOnSharedPreferenceChangeListener(this);
+    updateSummaryB(sharedPreferences, keyprefVideoCall);
+    updateSummaryB(sharedPreferences, keyprefScreencapture);
+    updateSummaryB(sharedPreferences, keyprefCamera2);
+    updateSummary(sharedPreferences, keyprefResolution);
+    updateSummary(sharedPreferences, keyprefFps);
+    updateSummaryB(sharedPreferences, keyprefCaptureQualitySlider);
+    updateSummary(sharedPreferences, keyprefMaxVideoBitrateType);
+    updateSummaryBitrate(sharedPreferences, keyprefMaxVideoBitrateValue);
+    setVideoBitrateEnable(sharedPreferences);
+    updateSummary(sharedPreferences, keyPrefVideoCodec);
+    updateSummaryB(sharedPreferences, keyprefHwCodec);
+    updateSummaryB(sharedPreferences, keyprefCaptureToTexture);
+    updateSummaryB(sharedPreferences, keyprefFlexfec);
+
+    updateSummary(sharedPreferences, keyprefStartAudioBitrateType);
+    updateSummaryBitrate(sharedPreferences, keyprefStartAudioBitrateValue);
+    setAudioBitrateEnable(sharedPreferences);
+    updateSummary(sharedPreferences, keyPrefAudioCodec);
+    updateSummaryB(sharedPreferences, keyprefNoAudioProcessing);
+    updateSummaryB(sharedPreferences, keyprefAecDump);
+    updateSummaryB(sharedPreferences, keyprefEnableSaveInputAudioToFile);
+    updateSummaryB(sharedPreferences, keyprefOpenSLES);
+    updateSummaryB(sharedPreferences, keyprefDisableBuiltInAEC);
+    updateSummaryB(sharedPreferences, keyprefDisableBuiltInAGC);
+    updateSummaryB(sharedPreferences, keyprefDisableBuiltInNS);
+    updateSummaryB(sharedPreferences, keyprefDisableWebRtcAGCAndHPF);
+    updateSummaryList(sharedPreferences, keyprefSpeakerphone);
+
+    updateSummaryB(sharedPreferences, keyprefEnableDataChannel);
+    updateSummaryB(sharedPreferences, keyprefOrdered);
+    updateSummary(sharedPreferences, keyprefMaxRetransmitTimeMs);
+    updateSummary(sharedPreferences, keyprefMaxRetransmits);
+    updateSummary(sharedPreferences, keyprefDataProtocol);
+    updateSummaryB(sharedPreferences, keyprefNegotiated);
+    updateSummary(sharedPreferences, keyprefDataId);
+    setDataChannelEnable(sharedPreferences);
+
+    updateSummary(sharedPreferences, keyPrefRoomServerUrl);
+    updateSummaryB(sharedPreferences, keyPrefDisplayHud);
+    updateSummaryB(sharedPreferences, keyPrefTracing);
+    updateSummaryB(sharedPreferences, keyprefEnabledRtcEventLog);
+
+    if (!Camera2Enumerator.isSupported(this)) {
+      Preference camera2Preference = settingsFragment.findPreference(keyprefCamera2);
+
+      camera2Preference.setSummary(getString(R.string.pref_camera2_not_supported));
+      camera2Preference.setEnabled(false);
+    }
+
+    if (!JavaAudioDeviceModule.isBuiltInAcousticEchoCancelerSupported()) {
+      Preference disableBuiltInAECPreference =
+          settingsFragment.findPreference(keyprefDisableBuiltInAEC);
+
+      disableBuiltInAECPreference.setSummary(getString(R.string.pref_built_in_aec_not_available));
+      disableBuiltInAECPreference.setEnabled(false);
+    }
+
+    Preference disableBuiltInAGCPreference =
+        settingsFragment.findPreference(keyprefDisableBuiltInAGC);
+
+    disableBuiltInAGCPreference.setSummary(getString(R.string.pref_built_in_agc_not_available));
+    disableBuiltInAGCPreference.setEnabled(false);
+
+    if (!JavaAudioDeviceModule.isBuiltInNoiseSuppressorSupported()) {
+      Preference disableBuiltInNSPreference =
+          settingsFragment.findPreference(keyprefDisableBuiltInNS);
+
+      disableBuiltInNSPreference.setSummary(getString(R.string.pref_built_in_ns_not_available));
+      disableBuiltInNSPreference.setEnabled(false);
+    }
+  }
+
+  @Override
+  protected void onPause() {
+    super.onPause();
+    SharedPreferences sharedPreferences =
+        settingsFragment.getPreferenceScreen().getSharedPreferences();
+    sharedPreferences.unregisterOnSharedPreferenceChangeListener(this);
+  }
+
+  @Override
+  public void onSharedPreferenceChanged(SharedPreferences sharedPreferences, String key) {
+    // clang-format off
+    if (key.equals(keyprefResolution)
+        || key.equals(keyprefFps)
+        || key.equals(keyprefMaxVideoBitrateType)
+        || key.equals(keyPrefVideoCodec)
+        || key.equals(keyprefStartAudioBitrateType)
+        || key.equals(keyPrefAudioCodec)
+        || key.equals(keyPrefRoomServerUrl)
+        || key.equals(keyprefMaxRetransmitTimeMs)
+        || key.equals(keyprefMaxRetransmits)
+        || key.equals(keyprefDataProtocol)
+        || key.equals(keyprefDataId)) {
+      updateSummary(sharedPreferences, key);
+    } else if (key.equals(keyprefMaxVideoBitrateValue)
+        || key.equals(keyprefStartAudioBitrateValue)) {
+      updateSummaryBitrate(sharedPreferences, key);
+    } else if (key.equals(keyprefVideoCall)
+        || key.equals(keyprefScreencapture)
+        || key.equals(keyprefCamera2)
+        || key.equals(keyPrefTracing)
+        || key.equals(keyprefCaptureQualitySlider)
+        || key.equals(keyprefHwCodec)
+        || key.equals(keyprefCaptureToTexture)
+        || key.equals(keyprefFlexfec)
+        || key.equals(keyprefNoAudioProcessing)
+        || key.equals(keyprefAecDump)
+        || key.equals(keyprefEnableSaveInputAudioToFile)
+        || key.equals(keyprefOpenSLES)
+        || key.equals(keyprefDisableBuiltInAEC)
+        || key.equals(keyprefDisableBuiltInAGC)
+        || key.equals(keyprefDisableBuiltInNS)
+        || key.equals(keyprefDisableWebRtcAGCAndHPF)
+        || key.equals(keyPrefDisplayHud)
+        || key.equals(keyprefEnableDataChannel)
+        || key.equals(keyprefOrdered)
+        || key.equals(keyprefNegotiated)
+        || key.equals(keyprefEnabledRtcEventLog)) {
+      updateSummaryB(sharedPreferences, key);
+    } else if (key.equals(keyprefSpeakerphone)) {
+      updateSummaryList(sharedPreferences, key);
+    }
+    // clang-format on
+    if (key.equals(keyprefMaxVideoBitrateType)) {
+      setVideoBitrateEnable(sharedPreferences);
+    }
+    if (key.equals(keyprefStartAudioBitrateType)) {
+      setAudioBitrateEnable(sharedPreferences);
+    }
+    if (key.equals(keyprefEnableDataChannel)) {
+      setDataChannelEnable(sharedPreferences);
+    }
+  }
+
+  private void updateSummary(SharedPreferences sharedPreferences, String key) {
+    Preference updatedPref = settingsFragment.findPreference(key);
+    // Set summary to be the user-description for the selected value
+    updatedPref.setSummary(sharedPreferences.getString(key, ""));
+  }
+
+  private void updateSummaryBitrate(SharedPreferences sharedPreferences, String key) {
+    Preference updatedPref = settingsFragment.findPreference(key);
+    updatedPref.setSummary(sharedPreferences.getString(key, "") + " kbps");
+  }
+
+  private void updateSummaryB(SharedPreferences sharedPreferences, String key) {
+    Preference updatedPref = settingsFragment.findPreference(key);
+    updatedPref.setSummary(sharedPreferences.getBoolean(key, true)
+            ? getString(R.string.pref_value_enabled)
+            : getString(R.string.pref_value_disabled));
+  }
+
+  private void updateSummaryList(SharedPreferences sharedPreferences, String key) {
+    ListPreference updatedPref = (ListPreference) settingsFragment.findPreference(key);
+    updatedPref.setSummary(updatedPref.getEntry());
+  }
+
+  private void setVideoBitrateEnable(SharedPreferences sharedPreferences) {
+    Preference bitratePreferenceValue =
+        settingsFragment.findPreference(keyprefMaxVideoBitrateValue);
+    String bitrateTypeDefault = getString(R.string.pref_maxvideobitrate_default);
+    String bitrateType =
+        sharedPreferences.getString(keyprefMaxVideoBitrateType, bitrateTypeDefault);
+    if (bitrateType.equals(bitrateTypeDefault)) {
+      bitratePreferenceValue.setEnabled(false);
+    } else {
+      bitratePreferenceValue.setEnabled(true);
+    }
+  }
+
+  private void setAudioBitrateEnable(SharedPreferences sharedPreferences) {
+    Preference bitratePreferenceValue =
+        settingsFragment.findPreference(keyprefStartAudioBitrateValue);
+    String bitrateTypeDefault = getString(R.string.pref_startaudiobitrate_default);
+    String bitrateType =
+        sharedPreferences.getString(keyprefStartAudioBitrateType, bitrateTypeDefault);
+    if (bitrateType.equals(bitrateTypeDefault)) {
+      bitratePreferenceValue.setEnabled(false);
+    } else {
+      bitratePreferenceValue.setEnabled(true);
+    }
+  }
+
+  private void setDataChannelEnable(SharedPreferences sharedPreferences) {
+    boolean enabled = sharedPreferences.getBoolean(keyprefEnableDataChannel, true);
+    settingsFragment.findPreference(keyprefOrdered).setEnabled(enabled);
+    settingsFragment.findPreference(keyprefMaxRetransmitTimeMs).setEnabled(enabled);
+    settingsFragment.findPreference(keyprefMaxRetransmits).setEnabled(enabled);
+    settingsFragment.findPreference(keyprefDataProtocol).setEnabled(enabled);
+    settingsFragment.findPreference(keyprefNegotiated).setEnabled(enabled);
+    settingsFragment.findPreference(keyprefDataId).setEnabled(enabled);
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/SettingsFragment.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/SettingsFragment.java
new file mode 100644
index 0000000000..d969bd7d32
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/SettingsFragment.java
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.os.Bundle;
+import android.preference.PreferenceFragment;
+
+/**
+ * Settings fragment for AppRTC.
+ */
+public class SettingsFragment extends PreferenceFragment {
+  @Override
+  public void onCreate(Bundle savedInstanceState) {
+    super.onCreate(savedInstanceState);
+    // Load the preferences from an XML resource
+    addPreferencesFromResource(R.xml.preferences);
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/TCPChannelClient.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/TCPChannelClient.java
new file mode 100644
index 0000000000..d869d7ca66
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/TCPChannelClient.java
@@ -0,0 +1,362 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.util.Log;
+import androidx.annotation.Nullable;
+import java.io.BufferedReader;
+import java.io.IOException;
+import java.io.InputStreamReader;
+import java.io.OutputStreamWriter;
+import java.io.PrintWriter;
+import java.net.InetAddress;
+import java.net.ServerSocket;
+import java.net.Socket;
+import java.net.UnknownHostException;
+import java.nio.charset.Charset;
+import java.util.concurrent.ExecutorService;
+import org.webrtc.ThreadUtils;
+
+/**
+ * Replacement for WebSocketChannelClient for direct communication between two IP addresses. Handles
+ * the signaling between the two clients using a TCP connection.
+ * <p>
+ * All public methods should be called from a looper executor thread
+ * passed in a constructor, otherwise exception will be thrown.
+ * All events are dispatched on the same thread.
+ */
+public class TCPChannelClient {
+  private static final String TAG = "TCPChannelClient";
+
+  private final ExecutorService executor;
+  private final ThreadUtils.ThreadChecker executorThreadCheck;
+  private final TCPChannelEvents eventListener;
+  private TCPSocket socket;
+
+  /**
+   * Callback interface for messages delivered on TCP Connection. All callbacks are invoked from the
+   * looper executor thread.
+   */
+  public interface TCPChannelEvents {
+    void onTCPConnected(boolean server);
+    void onTCPMessage(String message);
+    void onTCPError(String description);
+    void onTCPClose();
+  }
+
+  /**
+   * Initializes the TCPChannelClient. If IP is a local IP address, starts a listening server on
+   * that IP. If not, instead connects to the IP.
+   *
+   * @param eventListener Listener that will receive events from the client.
+   * @param ip            IP address to listen on or connect to.
+   * @param port          Port to listen on or connect to.
+   */
+  public TCPChannelClient(
+      ExecutorService executor, TCPChannelEvents eventListener, String ip, int port) {
+    this.executor = executor;
+    executorThreadCheck = new ThreadUtils.ThreadChecker();
+    executorThreadCheck.detachThread();
+    this.eventListener = eventListener;
+
+    InetAddress address;
+    try {
+      address = InetAddress.getByName(ip);
+    } catch (UnknownHostException e) {
+      reportError("Invalid IP address.");
+      return;
+    }
+
+    if (address.isAnyLocalAddress()) {
+      socket = new TCPSocketServer(address, port);
+    } else {
+      socket = new TCPSocketClient(address, port);
+    }
+
+    socket.start();
+  }
+
+  /**
+   * Disconnects the client if not already disconnected. This will fire the onTCPClose event.
+   */
+  public void disconnect() {
+    executorThreadCheck.checkIsOnValidThread();
+
+    socket.disconnect();
+  }
+
+  /**
+   * Sends a message on the socket.
+   *
+   * @param message Message to be sent.
+   */
+  public void send(String message) {
+    executorThreadCheck.checkIsOnValidThread();
+
+    socket.send(message);
+  }
+
+  /**
+   * Helper method for firing onTCPError events. Calls onTCPError on the executor thread.
+   */
+  private void reportError(final String message) {
+    Log.e(TAG, "TCP Error: " + message);
+    executor.execute(new Runnable() {
+      @Override
+      public void run() {
+        eventListener.onTCPError(message);
+      }
+    });
+  }
+
+  /**
+   * Base class for server and client sockets. Contains a listening thread that will call
+   * eventListener.onTCPMessage on new messages.
+   */
+  private abstract class TCPSocket extends Thread {
+    // Lock for editing out and rawSocket
+    protected final Object rawSocketLock;
+    @Nullable
+    private PrintWriter out;
+    @Nullable
+    private Socket rawSocket;
+
+    /**
+     * Connect to the peer, potentially a slow operation.
+     *
+     * @return Socket connection, null if connection failed.
+     */
+    @Nullable
+    public abstract Socket connect();
+
+    /** Returns true if sockets is a server rawSocket. */
+    public abstract boolean isServer();
+
+    TCPSocket() {
+      rawSocketLock = new Object();
+    }
+
+    /**
+     * The listening thread.
+     */
+    @Override
+    public void run() {
+      Log.d(TAG, "Listening thread started...");
+
+      // Receive connection to temporary variable first, so we don't block.
+      Socket tempSocket = connect();
+      BufferedReader in;
+
+      Log.d(TAG, "TCP connection established.");
+
+      synchronized (rawSocketLock) {
+        if (rawSocket != null) {
+          Log.e(TAG, "Socket already existed and will be replaced.");
+        }
+
+        rawSocket = tempSocket;
+
+        // Connecting failed, error has already been reported, just exit.
+        if (rawSocket == null) {
+          return;
+        }
+
+        try {
+          out = new PrintWriter(
+              new OutputStreamWriter(rawSocket.getOutputStream(), Charset.forName("UTF-8")), true);
+          in = new BufferedReader(
+              new InputStreamReader(rawSocket.getInputStream(), Charset.forName("UTF-8")));
+        } catch (IOException e) {
+          reportError("Failed to open IO on rawSocket: " + e.getMessage());
+          return;
+        }
+      }
+
+      Log.v(TAG, "Execute onTCPConnected");
+      executor.execute(new Runnable() {
+        @Override
+        public void run() {
+          Log.v(TAG, "Run onTCPConnected");
+          eventListener.onTCPConnected(isServer());
+        }
+      });
+
+      while (true) {
+        final String message;
+        try {
+          message = in.readLine();
+        } catch (IOException e) {
+          synchronized (rawSocketLock) {
+            // If socket was closed, this is expected.
+            if (rawSocket == null) {
+              break;
+            }
+          }
+
+          reportError("Failed to read from rawSocket: " + e.getMessage());
+          break;
+        }
+
+        // No data received, rawSocket probably closed.
+        if (message == null) {
+          break;
+        }
+
+        executor.execute(new Runnable() {
+          @Override
+          public void run() {
+            Log.v(TAG, "Receive: " + message);
+            eventListener.onTCPMessage(message);
+          }
+        });
+      }
+
+      Log.d(TAG, "Receiving thread exiting...");
+
+      // Close the rawSocket if it is still open.
+      disconnect();
+    }
+
+    /** Closes the rawSocket if it is still open. Also fires the onTCPClose event. */
+    public void disconnect() {
+      try {
+        synchronized (rawSocketLock) {
+          if (rawSocket != null) {
+            rawSocket.close();
+            rawSocket = null;
+            out = null;
+
+            executor.execute(new Runnable() {
+              @Override
+              public void run() {
+                eventListener.onTCPClose();
+              }
+            });
+          }
+        }
+      } catch (IOException e) {
+        reportError("Failed to close rawSocket: " + e.getMessage());
+      }
+    }
+
+    /**
+     * Sends a message on the socket. Should only be called on the executor thread.
+     */
+    public void send(String message) {
+      Log.v(TAG, "Send: " + message);
+
+      synchronized (rawSocketLock) {
+        if (out == null) {
+          reportError("Sending data on closed socket.");
+          return;
+        }
+
+        out.write(message + "\n");
+        out.flush();
+      }
+    }
+  }
+
+  private class TCPSocketServer extends TCPSocket {
+    // Server socket is also guarded by rawSocketLock.
+    @Nullable
+    private ServerSocket serverSocket;
+
+    final private InetAddress address;
+    final private int port;
+
+    public TCPSocketServer(InetAddress address, int port) {
+      this.address = address;
+      this.port = port;
+    }
+
+    /** Opens a listening socket and waits for a connection. */
+    @Nullable
+    @Override
+    public Socket connect() {
+      Log.d(TAG, "Listening on [" + address.getHostAddress() + "]:" + Integer.toString(port));
+
+      final ServerSocket tempSocket;
+      try {
+        tempSocket = new ServerSocket(port, 0, address);
+      } catch (IOException e) {
+        reportError("Failed to create server socket: " + e.getMessage());
+        return null;
+      }
+
+      synchronized (rawSocketLock) {
+        if (serverSocket != null) {
+          Log.e(TAG, "Server rawSocket was already listening and new will be opened.");
+        }
+
+        serverSocket = tempSocket;
+      }
+
+      try {
+        return tempSocket.accept();
+      } catch (IOException e) {
+        reportError("Failed to receive connection: " + e.getMessage());
+        return null;
+      }
+    }
+
+    /** Closes the listening socket and calls super. */
+    @Override
+    public void disconnect() {
+      try {
+        synchronized (rawSocketLock) {
+          if (serverSocket != null) {
+            serverSocket.close();
+            serverSocket = null;
+          }
+        }
+      } catch (IOException e) {
+        reportError("Failed to close server socket: " + e.getMessage());
+      }
+
+      super.disconnect();
+    }
+
+    @Override
+    public boolean isServer() {
+      return true;
+    }
+  }
+
+  private class TCPSocketClient extends TCPSocket {
+    final private InetAddress address;
+    final private int port;
+
+    public TCPSocketClient(InetAddress address, int port) {
+      this.address = address;
+      this.port = port;
+    }
+
+    /** Connects to the peer. */
+    @Nullable
+    @Override
+    public Socket connect() {
+      Log.d(TAG, "Connecting to [" + address.getHostAddress() + "]:" + Integer.toString(port));
+
+      try {
+        return new Socket(address, port);
+      } catch (IOException e) {
+        reportError("Failed to connect: " + e.getMessage());
+        return null;
+      }
+    }
+
+    @Override
+    public boolean isServer() {
+      return false;
+    }
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/UnhandledExceptionHandler.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/UnhandledExceptionHandler.java
new file mode 100644
index 0000000000..b256400119
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/UnhandledExceptionHandler.java
@@ -0,0 +1,85 @@
+/*
+ *  Copyright 2013 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.app.Activity;
+import android.app.AlertDialog;
+import android.content.DialogInterface;
+import android.util.Log;
+import android.util.TypedValue;
+import android.widget.ScrollView;
+import android.widget.TextView;
+
+import java.io.PrintWriter;
+import java.io.StringWriter;
+
+/**
+ * Singleton helper: install a default unhandled exception handler which shows
+ * an informative dialog and kills the app.  Useful for apps whose
+ * error-handling consists of throwing RuntimeExceptions.
+ * NOTE: almost always more useful to
+ * Thread.setDefaultUncaughtExceptionHandler() rather than
+ * Thread.setUncaughtExceptionHandler(), to apply to background threads as well.
+ */
+public class UnhandledExceptionHandler implements Thread.UncaughtExceptionHandler {
+  private static final String TAG = "AppRTCMobileActivity";
+  private final Activity activity;
+
+  public UnhandledExceptionHandler(final Activity activity) {
+    this.activity = activity;
+  }
+
+  @Override
+  public void uncaughtException(Thread unusedThread, final Throwable e) {
+    activity.runOnUiThread(new Runnable() {
+      @Override
+      public void run() {
+        String title = "Fatal error: " + getTopLevelCauseMessage(e);
+        String msg = getRecursiveStackTrace(e);
+        TextView errorView = new TextView(activity);
+        errorView.setText(msg);
+        errorView.setTextSize(TypedValue.COMPLEX_UNIT_SP, 8);
+        ScrollView scrollingContainer = new ScrollView(activity);
+        scrollingContainer.addView(errorView);
+        Log.e(TAG, title + "\n\n" + msg);
+        DialogInterface.OnClickListener listener = new DialogInterface.OnClickListener() {
+          @Override
+          public void onClick(DialogInterface dialog, int which) {
+            dialog.dismiss();
+            System.exit(1);
+          }
+        };
+        AlertDialog.Builder builder = new AlertDialog.Builder(activity);
+        builder.setTitle(title)
+            .setView(scrollingContainer)
+            .setPositiveButton("Exit", listener)
+            .show();
+      }
+    });
+  }
+
+  // Returns the Message attached to the original Cause of `t`.
+  private static String getTopLevelCauseMessage(Throwable t) {
+    Throwable topLevelCause = t;
+    while (topLevelCause.getCause() != null) {
+      topLevelCause = topLevelCause.getCause();
+    }
+    return topLevelCause.getMessage();
+  }
+
+  // Returns a human-readable String of the stacktrace in `t`, recursively
+  // through all Causes that led to `t`.
+  private static String getRecursiveStackTrace(Throwable t) {
+    StringWriter writer = new StringWriter();
+    t.printStackTrace(new PrintWriter(writer));
+    return writer.toString();
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java
new file mode 100644
index 0000000000..5fa410889a
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java
@@ -0,0 +1,296 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.os.Handler;
+import android.util.Log;
+import androidx.annotation.Nullable;
+import de.tavendo.autobahn.WebSocket.WebSocketConnectionObserver;
+import de.tavendo.autobahn.WebSocketConnection;
+import de.tavendo.autobahn.WebSocketException;
+import java.net.URI;
+import java.net.URISyntaxException;
+import java.util.ArrayList;
+import java.util.List;
+import org.appspot.apprtc.util.AsyncHttpURLConnection;
+import org.appspot.apprtc.util.AsyncHttpURLConnection.AsyncHttpEvents;
+import org.json.JSONException;
+import org.json.JSONObject;
+
+/**
+ * WebSocket client implementation.
+ *
+ * <p>All public methods should be called from a looper executor thread
+ * passed in a constructor, otherwise exception will be thrown.
+ * All events are dispatched on the same thread.
+ */
+public class WebSocketChannelClient {
+  private static final String TAG = "WSChannelRTCClient";
+  private static final int CLOSE_TIMEOUT = 1000;
+  private final WebSocketChannelEvents events;
+  private final Handler handler;
+  private WebSocketConnection ws;
+  private String wsServerUrl;
+  private String postServerUrl;
+  @Nullable
+  private String roomID;
+  @Nullable
+  private String clientID;
+  private WebSocketConnectionState state;
+  // Do not remove this member variable. If this is removed, the observer gets garbage collected and
+  // this causes test breakages.
+  private WebSocketObserver wsObserver;
+  private final Object closeEventLock = new Object();
+  private boolean closeEvent;
+  // WebSocket send queue. Messages are added to the queue when WebSocket
+  // client is not registered and are consumed in register() call.
+  private final List<String> wsSendQueue = new ArrayList<>();
+
+  /**
+   * Possible WebSocket connection states.
+   */
+  public enum WebSocketConnectionState { NEW, CONNECTED, REGISTERED, CLOSED, ERROR }
+
+  /**
+   * Callback interface for messages delivered on WebSocket.
+   * All events are dispatched from a looper executor thread.
+   */
+  public interface WebSocketChannelEvents {
+    void onWebSocketMessage(final String message);
+    void onWebSocketClose();
+    void onWebSocketError(final String description);
+  }
+
+  public WebSocketChannelClient(Handler handler, WebSocketChannelEvents events) {
+    this.handler = handler;
+    this.events = events;
+    roomID = null;
+    clientID = null;
+    state = WebSocketConnectionState.NEW;
+  }
+
+  public WebSocketConnectionState getState() {
+    return state;
+  }
+
+  public void connect(final String wsUrl, final String postUrl) {
+    checkIfCalledOnValidThread();
+    if (state != WebSocketConnectionState.NEW) {
+      Log.e(TAG, "WebSocket is already connected.");
+      return;
+    }
+    wsServerUrl = wsUrl;
+    postServerUrl = postUrl;
+    closeEvent = false;
+
+    Log.d(TAG, "Connecting WebSocket to: " + wsUrl + ". Post URL: " + postUrl);
+    ws = new WebSocketConnection();
+    wsObserver = new WebSocketObserver();
+    try {
+      ws.connect(new URI(wsServerUrl), wsObserver);
+    } catch (URISyntaxException e) {
+      reportError("URI error: " + e.getMessage());
+    } catch (WebSocketException e) {
+      reportError("WebSocket connection error: " + e.getMessage());
+    }
+  }
+
+  public void register(final String roomID, final String clientID) {
+    checkIfCalledOnValidThread();
+    this.roomID = roomID;
+    this.clientID = clientID;
+    if (state != WebSocketConnectionState.CONNECTED) {
+      Log.w(TAG, "WebSocket register() in state " + state);
+      return;
+    }
+    Log.d(TAG, "Registering WebSocket for room " + roomID + ". ClientID: " + clientID);
+    JSONObject json = new JSONObject();
+    try {
+      json.put("cmd", "register");
+      json.put("roomid", roomID);
+      json.put("clientid", clientID);
+      Log.d(TAG, "C->WSS: " + json.toString());
+      ws.sendTextMessage(json.toString());
+      state = WebSocketConnectionState.REGISTERED;
+      // Send any previously accumulated messages.
+      for (String sendMessage : wsSendQueue) {
+        send(sendMessage);
+      }
+      wsSendQueue.clear();
+    } catch (JSONException e) {
+      reportError("WebSocket register JSON error: " + e.getMessage());
+    }
+  }
+
+  public void send(String message) {
+    checkIfCalledOnValidThread();
+    switch (state) {
+      case NEW:
+      case CONNECTED:
+        // Store outgoing messages and send them after websocket client
+        // is registered.
+        Log.d(TAG, "WS ACC: " + message);
+        wsSendQueue.add(message);
+        return;
+      case ERROR:
+      case CLOSED:
+        Log.e(TAG, "WebSocket send() in error or closed state : " + message);
+        return;
+      case REGISTERED:
+        JSONObject json = new JSONObject();
+        try {
+          json.put("cmd", "send");
+          json.put("msg", message);
+          message = json.toString();
+          Log.d(TAG, "C->WSS: " + message);
+          ws.sendTextMessage(message);
+        } catch (JSONException e) {
+          reportError("WebSocket send JSON error: " + e.getMessage());
+        }
+        break;
+    }
+  }
+
+  // This call can be used to send WebSocket messages before WebSocket
+  // connection is opened.
+  public void post(String message) {
+    checkIfCalledOnValidThread();
+    sendWSSMessage("POST", message);
+  }
+
+  public void disconnect(boolean waitForComplete) {
+    checkIfCalledOnValidThread();
+    Log.d(TAG, "Disconnect WebSocket. State: " + state);
+    if (state == WebSocketConnectionState.REGISTERED) {
+      // Send "bye" to WebSocket server.
+      send("{\"type\": \"bye\"}");
+      state = WebSocketConnectionState.CONNECTED;
+      // Send http DELETE to http WebSocket server.
+      sendWSSMessage("DELETE", "");
+    }
+    // Close WebSocket in CONNECTED or ERROR states only.
+    if (state == WebSocketConnectionState.CONNECTED || state == WebSocketConnectionState.ERROR) {
+      ws.disconnect();
+      state = WebSocketConnectionState.CLOSED;
+
+      // Wait for websocket close event to prevent websocket library from
+      // sending any pending messages to deleted looper thread.
+      if (waitForComplete) {
+        synchronized (closeEventLock) {
+          while (!closeEvent) {
+            try {
+              closeEventLock.wait(CLOSE_TIMEOUT);
+              break;
+            } catch (InterruptedException e) {
+              Log.e(TAG, "Wait error: " + e.toString());
+            }
+          }
+        }
+      }
+    }
+    Log.d(TAG, "Disconnecting WebSocket done.");
+  }
+
+  private void reportError(final String errorMessage) {
+    Log.e(TAG, errorMessage);
+    handler.post(new Runnable() {
+      @Override
+      public void run() {
+        if (state != WebSocketConnectionState.ERROR) {
+          state = WebSocketConnectionState.ERROR;
+          events.onWebSocketError(errorMessage);
+        }
+      }
+    });
+  }
+
+  // Asynchronously send POST/DELETE to WebSocket server.
+  private void sendWSSMessage(final String method, final String message) {
+    String postUrl = postServerUrl + "/" + roomID + "/" + clientID;
+    Log.d(TAG, "WS " + method + " : " + postUrl + " : " + message);
+    AsyncHttpURLConnection httpConnection =
+        new AsyncHttpURLConnection(method, postUrl, message, new AsyncHttpEvents() {
+          @Override
+          public void onHttpError(String errorMessage) {
+            reportError("WS " + method + " error: " + errorMessage);
+          }
+
+          @Override
+          public void onHttpComplete(String response) {}
+        });
+    httpConnection.send();
+  }
+
+  // Helper method for debugging purposes. Ensures that WebSocket method is
+  // called on a looper thread.
+  private void checkIfCalledOnValidThread() {
+    if (Thread.currentThread() != handler.getLooper().getThread()) {
+      throw new IllegalStateException("WebSocket method is not called on valid thread");
+    }
+  }
+
+  private class WebSocketObserver implements WebSocketConnectionObserver {
+    @Override
+    public void onOpen() {
+      Log.d(TAG, "WebSocket connection opened to: " + wsServerUrl);
+      handler.post(new Runnable() {
+        @Override
+        public void run() {
+          state = WebSocketConnectionState.CONNECTED;
+          // Check if we have pending register request.
+          if (roomID != null && clientID != null) {
+            register(roomID, clientID);
+          }
+        }
+      });
+    }
+
+    @Override
+    public void onClose(WebSocketCloseNotification code, String reason) {
+      Log.d(TAG, "WebSocket connection closed. Code: " + code + ". Reason: " + reason + ". State: "
+              + state);
+      synchronized (closeEventLock) {
+        closeEvent = true;
+        closeEventLock.notify();
+      }
+      handler.post(new Runnable() {
+        @Override
+        public void run() {
+          if (state != WebSocketConnectionState.CLOSED) {
+            state = WebSocketConnectionState.CLOSED;
+            events.onWebSocketClose();
+          }
+        }
+      });
+    }
+
+    @Override
+    public void onTextMessage(String payload) {
+      Log.d(TAG, "WSS->C: " + payload);
+      final String message = payload;
+      handler.post(new Runnable() {
+        @Override
+        public void run() {
+          if (state == WebSocketConnectionState.CONNECTED
+              || state == WebSocketConnectionState.REGISTERED) {
+            events.onWebSocketMessage(message);
+          }
+        }
+      });
+    }
+
+    @Override
+    public void onRawTextMessage(byte[] payload) {}
+
+    @Override
+    public void onBinaryMessage(byte[] payload) {}
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java
new file mode 100644
index 0000000000..cbfdb21c91
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java
@@ -0,0 +1,427 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import android.os.Handler;
+import android.os.HandlerThread;
+import android.util.Log;
+import androidx.annotation.Nullable;
+import org.appspot.apprtc.RoomParametersFetcher.RoomParametersFetcherEvents;
+import org.appspot.apprtc.WebSocketChannelClient.WebSocketChannelEvents;
+import org.appspot.apprtc.WebSocketChannelClient.WebSocketConnectionState;
+import org.appspot.apprtc.util.AsyncHttpURLConnection;
+import org.appspot.apprtc.util.AsyncHttpURLConnection.AsyncHttpEvents;
+import org.json.JSONArray;
+import org.json.JSONException;
+import org.json.JSONObject;
+import org.webrtc.IceCandidate;
+import org.webrtc.SessionDescription;
+
+/**
+ * Negotiates signaling for chatting with https://appr.tc "rooms".
+ * Uses the client<->server specifics of the apprtc AppEngine webapp.
+ *
+ * <p>To use: create an instance of this object (registering a message handler) and
+ * call connectToRoom().  Once room connection is established
+ * onConnectedToRoom() callback with room parameters is invoked.
+ * Messages to other party (with local Ice candidates and answer SDP) can
+ * be sent after WebSocket connection is established.
+ */
+public class WebSocketRTCClient implements AppRTCClient, WebSocketChannelEvents {
+  private static final String TAG = "WSRTCClient";
+  private static final String ROOM_JOIN = "join";
+  private static final String ROOM_MESSAGE = "message";
+  private static final String ROOM_LEAVE = "leave";
+
+  private enum ConnectionState { NEW, CONNECTED, CLOSED, ERROR }
+
+  private enum MessageType { MESSAGE, LEAVE }
+
+  private final Handler handler;
+  private boolean initiator;
+  private SignalingEvents events;
+  private WebSocketChannelClient wsClient;
+  private ConnectionState roomState;
+  private RoomConnectionParameters connectionParameters;
+  private String messageUrl;
+  private String leaveUrl;
+
+  public WebSocketRTCClient(SignalingEvents events) {
+    this.events = events;
+    roomState = ConnectionState.NEW;
+    final HandlerThread handlerThread = new HandlerThread(TAG);
+    handlerThread.start();
+    handler = new Handler(handlerThread.getLooper());
+  }
+
+  // --------------------------------------------------------------------
+  // AppRTCClient interface implementation.
+  // Asynchronously connect to an AppRTC room URL using supplied connection
+  // parameters, retrieves room parameters and connect to WebSocket server.
+  @Override
+  public void connectToRoom(RoomConnectionParameters connectionParameters) {
+    this.connectionParameters = connectionParameters;
+    handler.post(new Runnable() {
+      @Override
+      public void run() {
+        connectToRoomInternal();
+      }
+    });
+  }
+
+  @Override
+  public void disconnectFromRoom() {
+    handler.post(new Runnable() {
+      @Override
+      public void run() {
+        disconnectFromRoomInternal();
+        handler.getLooper().quit();
+      }
+    });
+  }
+
+  // Connects to room - function runs on a local looper thread.
+  private void connectToRoomInternal() {
+    String connectionUrl = getConnectionUrl(connectionParameters);
+    Log.d(TAG, "Connect to room: " + connectionUrl);
+    roomState = ConnectionState.NEW;
+    wsClient = new WebSocketChannelClient(handler, this);
+
+    RoomParametersFetcherEvents callbacks = new RoomParametersFetcherEvents() {
+      @Override
+      public void onSignalingParametersReady(final SignalingParameters params) {
+        WebSocketRTCClient.this.handler.post(new Runnable() {
+          @Override
+          public void run() {
+            WebSocketRTCClient.this.signalingParametersReady(params);
+          }
+        });
+      }
+
+      @Override
+      public void onSignalingParametersError(String description) {
+        WebSocketRTCClient.this.reportError(description);
+      }
+    };
+
+    new RoomParametersFetcher(connectionUrl, null, callbacks).makeRequest();
+  }
+
+  // Disconnect from room and send bye messages - runs on a local looper thread.
+  private void disconnectFromRoomInternal() {
+    Log.d(TAG, "Disconnect. Room state: " + roomState);
+    if (roomState == ConnectionState.CONNECTED) {
+      Log.d(TAG, "Closing room.");
+      sendPostMessage(MessageType.LEAVE, leaveUrl, null);
+    }
+    roomState = ConnectionState.CLOSED;
+    if (wsClient != null) {
+      wsClient.disconnect(true);
+    }
+  }
+
+  // Helper functions to get connection, post message and leave message URLs
+  private String getConnectionUrl(RoomConnectionParameters connectionParameters) {
+    return connectionParameters.roomUrl + "/" + ROOM_JOIN + "/" + connectionParameters.roomId
+        + getQueryString(connectionParameters);
+  }
+
+  private String getMessageUrl(
+      RoomConnectionParameters connectionParameters, SignalingParameters signalingParameters) {
+    return connectionParameters.roomUrl + "/" + ROOM_MESSAGE + "/" + connectionParameters.roomId
+        + "/" + signalingParameters.clientId + getQueryString(connectionParameters);
+  }
+
+  private String getLeaveUrl(
+      RoomConnectionParameters connectionParameters, SignalingParameters signalingParameters) {
+    return connectionParameters.roomUrl + "/" + ROOM_LEAVE + "/" + connectionParameters.roomId + "/"
+        + signalingParameters.clientId + getQueryString(connectionParameters);
+  }
+
+  private String getQueryString(RoomConnectionParameters connectionParameters) {
+    if (connectionParameters.urlParameters != null) {
+      return "?" + connectionParameters.urlParameters;
+    } else {
+      return "";
+    }
+  }
+
+  // Callback issued when room parameters are extracted. Runs on local
+  // looper thread.
+  private void signalingParametersReady(final SignalingParameters signalingParameters) {
+    Log.d(TAG, "Room connection completed.");
+    if (connectionParameters.loopback
+        && (!signalingParameters.initiator || signalingParameters.offerSdp != null)) {
+      reportError("Loopback room is busy.");
+      return;
+    }
+    if (!connectionParameters.loopback && !signalingParameters.initiator
+        && signalingParameters.offerSdp == null) {
+      Log.w(TAG, "No offer SDP in room response.");
+    }
+    initiator = signalingParameters.initiator;
+    messageUrl = getMessageUrl(connectionParameters, signalingParameters);
+    leaveUrl = getLeaveUrl(connectionParameters, signalingParameters);
+    Log.d(TAG, "Message URL: " + messageUrl);
+    Log.d(TAG, "Leave URL: " + leaveUrl);
+    roomState = ConnectionState.CONNECTED;
+
+    // Fire connection and signaling parameters events.
+    events.onConnectedToRoom(signalingParameters);
+
+    // Connect and register WebSocket client.
+    wsClient.connect(signalingParameters.wssUrl, signalingParameters.wssPostUrl);
+    wsClient.register(connectionParameters.roomId, signalingParameters.clientId);
+  }
+
+  // Send local offer SDP to the other participant.
+  @Override
+  public void sendOfferSdp(final SessionDescription sdp) {
+    handler.post(new Runnable() {
+      @Override
+      public void run() {
+        if (roomState != ConnectionState.CONNECTED) {
+          reportError("Sending offer SDP in non connected state.");
+          return;
+        }
+        JSONObject json = new JSONObject();
+        jsonPut(json, "sdp", sdp.description);
+        jsonPut(json, "type", "offer");
+        sendPostMessage(MessageType.MESSAGE, messageUrl, json.toString());
+        if (connectionParameters.loopback) {
+          // In loopback mode rename this offer to answer and route it back.
+          SessionDescription sdpAnswer = new SessionDescription(
+              SessionDescription.Type.fromCanonicalForm("answer"), sdp.description);
+          events.onRemoteDescription(sdpAnswer);
+        }
+      }
+    });
+  }
+
+  // Send local answer SDP to the other participant.
+  @Override
+  public void sendAnswerSdp(final SessionDescription sdp) {
+    handler.post(new Runnable() {
+      @Override
+      public void run() {
+        if (connectionParameters.loopback) {
+          Log.e(TAG, "Sending answer in loopback mode.");
+          return;
+        }
+        JSONObject json = new JSONObject();
+        jsonPut(json, "sdp", sdp.description);
+        jsonPut(json, "type", "answer");
+        wsClient.send(json.toString());
+      }
+    });
+  }
+
+  // Send Ice candidate to the other participant.
+  @Override
+  public void sendLocalIceCandidate(final IceCandidate candidate) {
+    handler.post(new Runnable() {
+      @Override
+      public void run() {
+        JSONObject json = new JSONObject();
+        jsonPut(json, "type", "candidate");
+        jsonPut(json, "label", candidate.sdpMLineIndex);
+        jsonPut(json, "id", candidate.sdpMid);
+        jsonPut(json, "candidate", candidate.sdp);
+        if (initiator) {
+          // Call initiator sends ice candidates to GAE server.
+          if (roomState != ConnectionState.CONNECTED) {
+            reportError("Sending ICE candidate in non connected state.");
+            return;
+          }
+          sendPostMessage(MessageType.MESSAGE, messageUrl, json.toString());
+          if (connectionParameters.loopback) {
+            events.onRemoteIceCandidate(candidate);
+          }
+        } else {
+          // Call receiver sends ice candidates to websocket server.
+          wsClient.send(json.toString());
+        }
+      }
+    });
+  }
+
+  // Send removed Ice candidates to the other participant.
+  @Override
+  public void sendLocalIceCandidateRemovals(final IceCandidate[] candidates) {
+    handler.post(new Runnable() {
+      @Override
+      public void run() {
+        JSONObject json = new JSONObject();
+        jsonPut(json, "type", "remove-candidates");
+        JSONArray jsonArray = new JSONArray();
+        for (final IceCandidate candidate : candidates) {
+          jsonArray.put(toJsonCandidate(candidate));
+        }
+        jsonPut(json, "candidates", jsonArray);
+        if (initiator) {
+          // Call initiator sends ice candidates to GAE server.
+          if (roomState != ConnectionState.CONNECTED) {
+            reportError("Sending ICE candidate removals in non connected state.");
+            return;
+          }
+          sendPostMessage(MessageType.MESSAGE, messageUrl, json.toString());
+          if (connectionParameters.loopback) {
+            events.onRemoteIceCandidatesRemoved(candidates);
+          }
+        } else {
+          // Call receiver sends ice candidates to websocket server.
+          wsClient.send(json.toString());
+        }
+      }
+    });
+  }
+
+  // --------------------------------------------------------------------
+  // WebSocketChannelEvents interface implementation.
+  // All events are called by WebSocketChannelClient on a local looper thread
+  // (passed to WebSocket client constructor).
+  @Override
+  public void onWebSocketMessage(final String msg) {
+    if (wsClient.getState() != WebSocketConnectionState.REGISTERED) {
+      Log.e(TAG, "Got WebSocket message in non registered state.");
+      return;
+    }
+    try {
+      JSONObject json = new JSONObject(msg);
+      String msgText = json.getString("msg");
+      String errorText = json.optString("error");
+      if (msgText.length() > 0) {
+        json = new JSONObject(msgText);
+        String type = json.optString("type");
+        if (type.equals("candidate")) {
+          events.onRemoteIceCandidate(toJavaCandidate(json));
+        } else if (type.equals("remove-candidates")) {
+          JSONArray candidateArray = json.getJSONArray("candidates");
+          IceCandidate[] candidates = new IceCandidate[candidateArray.length()];
+          for (int i = 0; i < candidateArray.length(); ++i) {
+            candidates[i] = toJavaCandidate(candidateArray.getJSONObject(i));
+          }
+          events.onRemoteIceCandidatesRemoved(candidates);
+        } else if (type.equals("answer")) {
+          if (initiator) {
+            SessionDescription sdp = new SessionDescription(
+                SessionDescription.Type.fromCanonicalForm(type), json.getString("sdp"));
+            events.onRemoteDescription(sdp);
+          } else {
+            reportError("Received answer for call initiator: " + msg);
+          }
+        } else if (type.equals("offer")) {
+          if (!initiator) {
+            SessionDescription sdp = new SessionDescription(
+                SessionDescription.Type.fromCanonicalForm(type), json.getString("sdp"));
+            events.onRemoteDescription(sdp);
+          } else {
+            reportError("Received offer for call receiver: " + msg);
+          }
+        } else if (type.equals("bye")) {
+          events.onChannelClose();
+        } else {
+          reportError("Unexpected WebSocket message: " + msg);
+        }
+      } else {
+        if (errorText != null && errorText.length() > 0) {
+          reportError("WebSocket error message: " + errorText);
+        } else {
+          reportError("Unexpected WebSocket message: " + msg);
+        }
+      }
+    } catch (JSONException e) {
+      reportError("WebSocket message JSON parsing error: " + e.toString());
+    }
+  }
+
+  @Override
+  public void onWebSocketClose() {
+    events.onChannelClose();
+  }
+
+  @Override
+  public void onWebSocketError(String description) {
+    reportError("WebSocket error: " + description);
+  }
+
+  // --------------------------------------------------------------------
+  // Helper functions.
+  private void reportError(final String errorMessage) {
+    Log.e(TAG, errorMessage);
+    handler.post(new Runnable() {
+      @Override
+      public void run() {
+        if (roomState != ConnectionState.ERROR) {
+          roomState = ConnectionState.ERROR;
+          events.onChannelError(errorMessage);
+        }
+      }
+    });
+  }
+
+  // Put a `key`->`value` mapping in `json`.
+  private static void jsonPut(JSONObject json, String key, Object value) {
+    try {
+      json.put(key, value);
+    } catch (JSONException e) {
+      throw new RuntimeException(e);
+    }
+  }
+
+  // Send SDP or ICE candidate to a room server.
+  private void sendPostMessage(
+      final MessageType messageType, final String url, @Nullable final String message) {
+    String logInfo = url;
+    if (message != null) {
+      logInfo += ". Message: " + message;
+    }
+    Log.d(TAG, "C->GAE: " + logInfo);
+    AsyncHttpURLConnection httpConnection =
+        new AsyncHttpURLConnection("POST", url, message, new AsyncHttpEvents() {
+          @Override
+          public void onHttpError(String errorMessage) {
+            reportError("GAE POST error: " + errorMessage);
+          }
+
+          @Override
+          public void onHttpComplete(String response) {
+            if (messageType == MessageType.MESSAGE) {
+              try {
+                JSONObject roomJson = new JSONObject(response);
+                String result = roomJson.getString("result");
+                if (!result.equals("SUCCESS")) {
+                  reportError("GAE POST error: " + result);
+                }
+              } catch (JSONException e) {
+                reportError("GAE POST JSON error: " + e.toString());
+              }
+            }
+          }
+        });
+    httpConnection.send();
+  }
+
+  // Converts a Java candidate to a JSONObject.
+  private JSONObject toJsonCandidate(final IceCandidate candidate) {
+    JSONObject json = new JSONObject();
+    jsonPut(json, "label", candidate.sdpMLineIndex);
+    jsonPut(json, "id", candidate.sdpMid);
+    jsonPut(json, "candidate", candidate.sdp);
+    return json;
+  }
+
+  // Converts a JSON candidate to a Java object.
+  IceCandidate toJavaCandidate(JSONObject json) throws JSONException {
+    return new IceCandidate(
+        json.getString("id"), json.getInt("label"), json.getString("candidate"));
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/util/AppRTCUtils.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/util/AppRTCUtils.java
new file mode 100644
index 0000000000..ee7f8c0416
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/util/AppRTCUtils.java
@@ -0,0 +1,47 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc.util;
+
+import android.os.Build;
+import android.util.Log;
+
+/**
+ * AppRTCUtils provides helper functions for managing thread safety.
+ */
+public final class AppRTCUtils {
+  private AppRTCUtils() {}
+
+  /** Helper method which throws an exception  when an assertion has failed. */
+  public static void assertIsTrue(boolean condition) {
+    if (!condition) {
+      throw new AssertionError("Expected condition to be true");
+    }
+  }
+
+  /** Helper method for building a string of thread information.*/
+  public static String getThreadInfo() {
+    return "@[name=" + Thread.currentThread().getName() + ", id=" + Thread.currentThread().getId()
+        + "]";
+  }
+
+  /** Information about the current build, taken from system properties. */
+  public static void logDeviceInfo(String tag) {
+    Log.d(tag, "Android SDK: " + Build.VERSION.SDK_INT + ", "
+            + "Release: " + Build.VERSION.RELEASE + ", "
+            + "Brand: " + Build.BRAND + ", "
+            + "Device: " + Build.DEVICE + ", "
+            + "Id: " + Build.ID + ", "
+            + "Hardware: " + Build.HARDWARE + ", "
+            + "Manufacturer: " + Build.MANUFACTURER + ", "
+            + "Model: " + Build.MODEL + ", "
+            + "Product: " + Build.PRODUCT);
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java
new file mode 100644
index 0000000000..93028ae783
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java
@@ -0,0 +1,115 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc.util;
+
+import java.io.IOException;
+import java.io.InputStream;
+import java.io.OutputStream;
+import java.net.HttpURLConnection;
+import java.net.SocketTimeoutException;
+import java.net.URL;
+import java.util.Scanner;
+
+/**
+ * Asynchronous http requests implementation.
+ */
+public class AsyncHttpURLConnection {
+  private static final int HTTP_TIMEOUT_MS = 8000;
+  private static final String HTTP_ORIGIN = "https://appr.tc";
+  private final String method;
+  private final String url;
+  private final String message;
+  private final AsyncHttpEvents events;
+  private String contentType;
+
+  /**
+   * Http requests callbacks.
+   */
+  public interface AsyncHttpEvents {
+    void onHttpError(String errorMessage);
+    void onHttpComplete(String response);
+  }
+
+  public AsyncHttpURLConnection(String method, String url, String message, AsyncHttpEvents events) {
+    this.method = method;
+    this.url = url;
+    this.message = message;
+    this.events = events;
+  }
+
+  public void setContentType(String contentType) {
+    this.contentType = contentType;
+  }
+
+  public void send() {
+    new Thread(this ::sendHttpMessage).start();
+  }
+
+  @SuppressWarnings("UseNetworkAnnotations")
+  private void sendHttpMessage() {
+    try {
+      HttpURLConnection connection = (HttpURLConnection) new URL(url).openConnection();
+      byte[] postData = new byte[0];
+      if (message != null) {
+        postData = message.getBytes("UTF-8");
+      }
+      connection.setRequestMethod(method);
+      connection.setUseCaches(false);
+      connection.setDoInput(true);
+      connection.setConnectTimeout(HTTP_TIMEOUT_MS);
+      connection.setReadTimeout(HTTP_TIMEOUT_MS);
+      // TODO(glaznev) - query request origin from pref_room_server_url_key preferences.
+      connection.addRequestProperty("origin", HTTP_ORIGIN);
+      boolean doOutput = false;
+      if (method.equals("POST")) {
+        doOutput = true;
+        connection.setDoOutput(true);
+        connection.setFixedLengthStreamingMode(postData.length);
+      }
+      if (contentType == null) {
+        connection.setRequestProperty("Content-Type", "text/plain; charset=utf-8");
+      } else {
+        connection.setRequestProperty("Content-Type", contentType);
+      }
+
+      // Send POST request.
+      if (doOutput && postData.length > 0) {
+        OutputStream outStream = connection.getOutputStream();
+        outStream.write(postData);
+        outStream.close();
+      }
+
+      // Get response.
+      int responseCode = connection.getResponseCode();
+      if (responseCode != 200) {
+        events.onHttpError("Non-200 response to " + method + " to URL: " + url + " : "
+            + connection.getHeaderField(null));
+        connection.disconnect();
+        return;
+      }
+      InputStream responseStream = connection.getInputStream();
+      String response = drainStream(responseStream);
+      responseStream.close();
+      connection.disconnect();
+      events.onHttpComplete(response);
+    } catch (SocketTimeoutException e) {
+      events.onHttpError("HTTP " + method + " to " + url + " timeout");
+    } catch (IOException e) {
+      events.onHttpError("HTTP " + method + " to " + url + " error: " + e.getMessage());
+    }
+  }
+
+  // Return the contents of an InputStream as a String.
+  private static String drainStream(InputStream in) {
+    Scanner s = new Scanner(in, "UTF-8").useDelimiter("\\A");
+    return s.hasNext() ? s.next() : "";
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/start_loopback_stubbed_camera_saved_video_out.py b/third_party/libwebrtc/examples/androidapp/start_loopback_stubbed_camera_saved_video_out.py
new file mode 100644
index 0000000000..b1cf84611f
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/start_loopback_stubbed_camera_saved_video_out.py
@@ -0,0 +1,127 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+from optparse import OptionParser
+import random
+import string
+import subprocess
+import sys
+import time
+
+from com.android.monkeyrunner import MonkeyRunner, MonkeyDevice
+
+
+def main():
+    parser = OptionParser()
+
+    parser.add_option('--devname', dest='devname', help='The device id')
+
+    parser.add_option(
+        '--videooutsave',
+        dest='videooutsave',
+        help='The path where to save the video out file on local computer')
+
+    parser.add_option('--videoout',
+                      dest='videoout',
+                      help='The path where to put the video out file')
+
+    parser.add_option('--videoout_width',
+                      dest='videoout_width',
+                      type='int',
+                      help='The width for the video out file')
+
+    parser.add_option('--videoout_height',
+                      dest='videoout_height',
+                      type='int',
+                      help='The height for the video out file')
+
+    parser.add_option(
+        '--videoin',
+        dest='videoin',
+        help='The path where to read input file instead of camera')
+
+    parser.add_option('--call_length',
+                      dest='call_length',
+                      type='int',
+                      help='The length of the call')
+
+    (options, args) = parser.parse_args()
+
+    print(options, args)
+
+    devname = options.devname
+
+    videoin = options.videoin
+
+    videoout = options.videoout
+    videoout_width = options.videoout_width
+    videoout_height = options.videoout_height
+
+    videooutsave = options.videooutsave
+
+    call_length = options.call_length or 10
+
+    room = ''.join(
+        random.choice(string.ascii_letters + string.digits) for _ in range(8))
+
+    # Delete output video file.
+    if videoout:
+        subprocess.check_call(
+            ['adb', '-s', devname, 'shell', 'rm', '-f', videoout])
+
+    device = MonkeyRunner.waitForConnection(2, devname)
+
+    extras = {
+        'org.appspot.apprtc.USE_VALUES_FROM_INTENT': True,
+        'org.appspot.apprtc.AUDIOCODEC': 'OPUS',
+        'org.appspot.apprtc.LOOPBACK': True,
+        'org.appspot.apprtc.VIDEOCODEC': 'VP8',
+        'org.appspot.apprtc.CAPTURETOTEXTURE': False,
+        'org.appspot.apprtc.CAMERA2': False,
+        'org.appspot.apprtc.ROOMID': room
+    }
+
+    if videoin:
+        extras.update({'org.appspot.apprtc.VIDEO_FILE_AS_CAMERA': videoin})
+
+    if videoout:
+        extras.update({
+            'org.appspot.apprtc.SAVE_REMOTE_VIDEO_TO_FILE':
+            videoout,
+            'org.appspot.apprtc.SAVE_REMOTE_VIDEO_TO_FILE_WIDTH':
+            videoout_width,
+            'org.appspot.apprtc.SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT':
+            videoout_height
+        })
+
+    print extras
+
+    device.startActivity(data='https://appr.tc',
+                         action='android.intent.action.VIEW',
+                         component='org.appspot.apprtc/.ConnectActivity',
+                         extras=extras)
+
+    print 'Running a call for %d seconds' % call_length
+    for _ in xrange(call_length):
+        sys.stdout.write('.')
+        sys.stdout.flush()
+        time.sleep(1)
+    print '\nEnding call.'
+
+    # Press back to end the call. Will end on both sides.
+    device.press('KEYCODE_BACK', MonkeyDevice.DOWN_AND_UP)
+
+    if videooutsave:
+        time.sleep(2)
+
+        subprocess.check_call(
+            ['adb', '-s', devname, 'pull', videoout, videooutsave])
+
+
+if __name__ == '__main__':
+    main()
diff --git a/third_party/libwebrtc/examples/androidapp/third_party/autobanh/BUILD.gn b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/BUILD.gn
new file mode 100644
index 0000000000..b671239bae
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/BUILD.gn
@@ -0,0 +1,15 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+if (is_android) {
+  import("//build/config/android/rules.gni")
+
+  android_java_prebuilt("autobanh_java") {
+    jar_path = "lib/autobanh.jar"
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidapp/third_party/autobanh/LICENSE b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/LICENSE
new file mode 100644
index 0000000000..f433b1a53f
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/LICENSE
@@ -0,0 +1,177 @@
+
+                                 Apache License
+                           Version 2.0, January 2004
+                        http://www.apache.org/licenses/
+
+   TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
+
+   1. Definitions.
+
+      "License" shall mean the terms and conditions for use, reproduction,
+      and distribution as defined by Sections 1 through 9 of this document.
+
+      "Licensor" shall mean the copyright owner or entity authorized by
+      the copyright owner that is granting the License.
+
+      "Legal Entity" shall mean the union of the acting entity and all
+      other entities that control, are controlled by, or are under common
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+      direction or management of such entity, whether by contract or
+      otherwise, or (ii) ownership of fifty percent (50%) or more of the
+      outstanding shares, or (iii) beneficial ownership of such entity.
+
+      "You" (or "Your") shall mean an individual or Legal Entity
+      exercising permissions granted by this License.
+
+      "Source" form shall mean the preferred form for making modifications,
+      including but not limited to software source code, documentation
+      source, and configuration files.
+
+      "Object" form shall mean any form resulting from mechanical
+      transformation or translation of a Source form, including but
+      not limited to compiled object code, generated documentation,
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+
+      "Work" shall mean the work of authorship, whether in Source or
+      Object form, made available under the License, as indicated by a
+      copyright notice that is included in or attached to the work
+      (an example is provided in the Appendix below).
+
+      "Derivative Works" shall mean any work, whether in Source or Object
+      form, that is based on (or derived from) the Work and for which the
+      editorial revisions, annotations, elaborations, or other modifications
+      represent, as a whole, an original work of authorship. For the purposes
+      of this License, Derivative Works shall not include works that remain
+      separable from, or merely link (or bind by name) to the interfaces of,
+      the Work and Derivative Works thereof.
+
+      "Contribution" shall mean any work of authorship, including
+      the original version of the Work and any modifications or additions
+      to that Work or Derivative Works thereof, that is intentionally
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+      communication on electronic mailing lists, source code control systems,
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+      excluding communication that is conspicuously marked or otherwise
+      designated in writing by the copyright owner as "Not a Contribution."
+
+      "Contributor" shall mean Licensor and any individual or Legal Entity
+      on behalf of whom a Contribution has been received by Licensor and
+      subsequently incorporated within the Work.
+
+   2. Grant of Copyright License. Subject to the terms and conditions of
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+          wherever such third-party notices normally appear. The contents
+          of the NOTICE file are for informational purposes only and
+          do not modify the License. You may add Your own attribution
+          notices within Derivative Works that You distribute, alongside
+          or as an addendum to the NOTICE text from the Work, provided
+          that such additional attribution notices cannot be construed
+          as modifying the License.
+
+      You may add Your own copyright statement to Your modifications and
+      may provide additional or different license terms and conditions
+      for use, reproduction, or distribution of Your modifications, or
+      for any such Derivative Works as a whole, provided Your use,
+      reproduction, and distribution of the Work otherwise complies with
+      the conditions stated in this License.
+
+   5. Submission of Contributions. Unless You explicitly state otherwise,
+      any Contribution intentionally submitted for inclusion in the Work
+      by You to the Licensor shall be under the terms and conditions of
+      this License, without any additional terms or conditions.
+      Notwithstanding the above, nothing herein shall supersede or modify
+      the terms of any separate license agreement you may have executed
+      with Licensor regarding such Contributions.
+
+   6. Trademarks. This License does not grant permission to use the trade
+      names, trademarks, service marks, or product names of the Licensor,
+      except as required for reasonable and customary use in describing the
+      origin of the Work and reproducing the content of the NOTICE file.
+
+   7. Disclaimer of Warranty. Unless required by applicable law or
+      agreed to in writing, Licensor provides the Work (and each
+      Contributor provides its Contributions) on an "AS IS" BASIS,
+      WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or
+      implied, including, without limitation, any warranties or conditions
+      of TITLE, NON-INFRINGEMENT, MERCHANTABILITY, or FITNESS FOR A
+      PARTICULAR PURPOSE. You are solely responsible for determining the
+      appropriateness of using or redistributing the Work and assume any
+      risks associated with Your exercise of permissions under this License.
+
+   8. Limitation of Liability. In no event and under no legal theory,
+      whether in tort (including negligence), contract, or otherwise,
+      unless required by applicable law (such as deliberate and grossly
+      negligent acts) or agreed to in writing, shall any Contributor be
+      liable to You for damages, including any direct, indirect, special,
+      incidental, or consequential damages of any character arising as a
+      result of this License or out of the use or inability to use the
+      Work (including but not limited to damages for loss of goodwill,
+      work stoppage, computer failure or malfunction, or any and all
+      other commercial damages or losses), even if such Contributor
+      has been advised of the possibility of such damages.
+
+   9. Accepting Warranty or Additional Liability. While redistributing
+      the Work or Derivative Works thereof, You may choose to offer,
+      and charge a fee for, acceptance of support, warranty, indemnity,
+      or other liability obligations and/or rights consistent with this
+      License. However, in accepting such obligations, You may act only
+      on Your own behalf and on Your sole responsibility, not on behalf
+      of any other Contributor, and only if You agree to indemnify,
+      defend, and hold each Contributor harmless for any liability
+      incurred by, or claims asserted against, such Contributor by reason
+      of your accepting any such warranty or additional liability.
+
+   END OF TERMS AND CONDITIONS
diff --git a/third_party/libwebrtc/examples/androidapp/third_party/autobanh/LICENSE.md b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/LICENSE.md
new file mode 100644
index 0000000000..2079e90d6b
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/LICENSE.md
@@ -0,0 +1,21 @@
+The MIT License (MIT)
+
+Copyright (c) 2014 Cameron Lowell Palmer
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.
diff --git a/third_party/libwebrtc/examples/androidapp/third_party/autobanh/NOTICE b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/NOTICE
new file mode 100644
index 0000000000..91ed7dfe0e
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/NOTICE
@@ -0,0 +1,3 @@
+AutobahnAndroid
+Copyright 2011,2012 Tavendo GmbH. Licensed under Apache 2.0
+This product includes software developed at Tavendo GmbH http://www.tavendo.de
diff --git a/third_party/libwebrtc/examples/androidapp/third_party/autobanh/lib/autobanh.jar b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/lib/autobanh.jar
new file mode 100644
index 0000000000..5a10b7f3f1
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidapp/third_party/autobanh/lib/autobanh.jar
diff --git a/third_party/libwebrtc/examples/androidjunit/OWNERS b/third_party/libwebrtc/examples/androidjunit/OWNERS
new file mode 100644
index 0000000000..cf092a316a
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidjunit/OWNERS
@@ -0,0 +1 @@
+xalep@webrtc.org
diff --git a/third_party/libwebrtc/examples/androidjunit/README b/third_party/libwebrtc/examples/androidjunit/README
new file mode 100644
index 0000000000..03902a779c
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidjunit/README
@@ -0,0 +1,8 @@
+This directory contains example JUnit tests for Android AppRTCMobile.
+Many of these test utilize Robolectric to mock Android classes.
+
+To compile:
+ninja -C out/Debug android_examples_junit_tests
+
+To run:
+out/Debug/bin/run_android_examples_junit_tests
diff --git a/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java b/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java
new file mode 100644
index 0000000000..3060bd7a56
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java
@@ -0,0 +1,268 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import static org.junit.Assert.assertEquals;
+import static org.junit.Assert.assertNotNull;
+import static org.junit.Assert.assertNull;
+import static org.mockito.Mockito.mock;
+import static org.mockito.Mockito.never;
+import static org.mockito.Mockito.times;
+import static org.mockito.Mockito.verify;
+import static org.mockito.Mockito.when;
+
+import android.bluetooth.BluetoothAdapter;
+import android.bluetooth.BluetoothDevice;
+import android.bluetooth.BluetoothHeadset;
+import android.bluetooth.BluetoothProfile;
+import android.content.BroadcastReceiver;
+import android.content.Context;
+import android.content.Intent;
+import android.content.IntentFilter;
+import android.media.AudioManager;
+import android.util.Log;
+import androidx.test.core.app.ApplicationProvider;
+import java.util.ArrayList;
+import java.util.List;
+import org.appspot.apprtc.AppRTCBluetoothManager.State;
+import org.chromium.testing.local.LocalRobolectricTestRunner;
+import org.junit.Before;
+import org.junit.Test;
+import org.junit.runner.RunWith;
+import org.robolectric.annotation.Config;
+import org.robolectric.shadows.ShadowLog;
+
+/**
+ * Verifies basic behavior of the AppRTCBluetoothManager class.
+ * Note that the test object uses an AppRTCAudioManager (injected in ctor),
+ * but a mocked version is used instead. Hence, the parts "driven" by the AppRTC
+ * audio manager are not included in this test.
+ */
+@RunWith(LocalRobolectricTestRunner.class)
+@Config(manifest = Config.NONE)
+public class BluetoothManagerTest {
+  private static final String TAG = "BluetoothManagerTest";
+  private static final String BLUETOOTH_TEST_DEVICE_NAME = "BluetoothTestDevice";
+
+  private BroadcastReceiver bluetoothHeadsetStateReceiver;
+  private BluetoothProfile.ServiceListener bluetoothServiceListener;
+  private BluetoothHeadset mockedBluetoothHeadset;
+  private BluetoothDevice mockedBluetoothDevice;
+  private List<BluetoothDevice> mockedBluetoothDeviceList;
+  private AppRTCBluetoothManager bluetoothManager;
+  private AppRTCAudioManager mockedAppRtcAudioManager;
+  private AudioManager mockedAudioManager;
+  private Context context;
+
+  @Before
+  public void setUp() {
+    ShadowLog.stream = System.out;
+    context = ApplicationProvider.getApplicationContext();
+    mockedAppRtcAudioManager = mock(AppRTCAudioManager.class);
+    mockedAudioManager = mock(AudioManager.class);
+    mockedBluetoothHeadset = mock(BluetoothHeadset.class);
+    mockedBluetoothDevice = mock(BluetoothDevice.class);
+    mockedBluetoothDeviceList = new ArrayList<BluetoothDevice>();
+
+    // Simulate that bluetooth SCO audio is available by default.
+    when(mockedAudioManager.isBluetoothScoAvailableOffCall()).thenReturn(true);
+
+    // Create the test object and override protected methods for this test.
+    bluetoothManager = new AppRTCBluetoothManager(context, mockedAppRtcAudioManager) {
+      @Override
+      protected AudioManager getAudioManager(Context context) {
+        Log.d(TAG, "getAudioManager");
+        return mockedAudioManager;
+      }
+
+      @Override
+      protected void registerReceiver(BroadcastReceiver receiver, IntentFilter filter) {
+        Log.d(TAG, "registerReceiver");
+        if (filter.hasAction(BluetoothHeadset.ACTION_CONNECTION_STATE_CHANGED)
+            && filter.hasAction(BluetoothHeadset.ACTION_AUDIO_STATE_CHANGED)) {
+          // Gives access to the real broadcast receiver so the test can use it.
+          bluetoothHeadsetStateReceiver = receiver;
+        }
+      }
+
+      @Override
+      protected void unregisterReceiver(BroadcastReceiver receiver) {
+        Log.d(TAG, "unregisterReceiver");
+        if (receiver == bluetoothHeadsetStateReceiver) {
+          bluetoothHeadsetStateReceiver = null;
+        }
+      }
+
+      @Override
+      protected boolean getBluetoothProfileProxy(
+          Context context, BluetoothProfile.ServiceListener listener, int profile) {
+        Log.d(TAG, "getBluetoothProfileProxy");
+        if (profile == BluetoothProfile.HEADSET) {
+          // Allows the test to access the real Bluetooth service listener object.
+          bluetoothServiceListener = listener;
+        }
+        return true;
+      }
+
+      @Override
+      protected boolean hasPermission(Context context, String permission) {
+        Log.d(TAG, "hasPermission(" + permission + ")");
+        // Ensure that the client asks for Bluetooth permission.
+        return android.Manifest.permission.BLUETOOTH.equals(permission);
+      }
+
+      @Override
+      protected void logBluetoothAdapterInfo(BluetoothAdapter localAdapter) {
+        // Do nothing in tests. No need to mock BluetoothAdapter.
+      }
+    };
+  }
+
+  // Verify that Bluetooth service listener for headset profile is properly initialized.
+  @Test
+  public void testBluetoothServiceListenerInitialized() {
+    bluetoothManager.start();
+    assertNotNull(bluetoothServiceListener);
+    verify(mockedAppRtcAudioManager, never()).updateAudioDeviceState();
+  }
+
+  // Verify that broadcast receivers for Bluetooth SCO audio state and Bluetooth headset state
+  // are properly registered and unregistered.
+  @Test
+  public void testBluetoothBroadcastReceiversAreRegistered() {
+    bluetoothManager.start();
+    assertNotNull(bluetoothHeadsetStateReceiver);
+    bluetoothManager.stop();
+    assertNull(bluetoothHeadsetStateReceiver);
+  }
+
+  // Verify that the Bluetooth manager starts and stops with correct states.
+  @Test
+  public void testBluetoothDefaultStartStopStates() {
+    bluetoothManager.start();
+    assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE);
+    bluetoothManager.stop();
+    assertEquals(bluetoothManager.getState(), State.UNINITIALIZED);
+  }
+
+  // Verify correct state after receiving BluetoothServiceListener.onServiceConnected()
+  // when no BT device is enabled.
+  @Test
+  public void testBluetoothServiceListenerConnectedWithNoHeadset() {
+    bluetoothManager.start();
+    assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE);
+    simulateBluetoothServiceConnectedWithNoConnectedHeadset();
+    verify(mockedAppRtcAudioManager, times(1)).updateAudioDeviceState();
+    assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE);
+  }
+
+  // Verify correct state after receiving BluetoothServiceListener.onServiceConnected()
+  // when one emulated (test) BT device is enabled. Android does not support more than
+  // one connected BT headset.
+  @Test
+  public void testBluetoothServiceListenerConnectedWithHeadset() {
+    bluetoothManager.start();
+    assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE);
+    simulateBluetoothServiceConnectedWithConnectedHeadset();
+    verify(mockedAppRtcAudioManager, times(1)).updateAudioDeviceState();
+    assertEquals(bluetoothManager.getState(), State.HEADSET_AVAILABLE);
+  }
+
+  // Verify correct state after receiving BluetoothProfile.ServiceListener.onServiceDisconnected().
+  @Test
+  public void testBluetoothServiceListenerDisconnected() {
+    bluetoothManager.start();
+    assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE);
+    simulateBluetoothServiceDisconnected();
+    verify(mockedAppRtcAudioManager, times(1)).updateAudioDeviceState();
+    assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE);
+  }
+
+  // Verify correct state after BluetoothServiceListener.onServiceConnected() and
+  // the intent indicating that the headset is actually connected. Both these callbacks
+  // results in calls to updateAudioDeviceState() on the AppRTC audio manager.
+  // No BT SCO is enabled here to keep the test limited.
+  @Test
+  public void testBluetoothHeadsetConnected() {
+    bluetoothManager.start();
+    assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE);
+    simulateBluetoothServiceConnectedWithConnectedHeadset();
+    simulateBluetoothHeadsetConnected();
+    verify(mockedAppRtcAudioManager, times(2)).updateAudioDeviceState();
+    assertEquals(bluetoothManager.getState(), State.HEADSET_AVAILABLE);
+  }
+
+  // Verify correct state sequence for a case when a BT headset is available,
+  // followed by BT SCO audio being enabled and then stopped.
+  @Test
+  public void testBluetoothScoAudioStartAndStop() {
+    bluetoothManager.start();
+    assertEquals(bluetoothManager.getState(), State.HEADSET_UNAVAILABLE);
+    simulateBluetoothServiceConnectedWithConnectedHeadset();
+    assertEquals(bluetoothManager.getState(), State.HEADSET_AVAILABLE);
+    bluetoothManager.startScoAudio();
+    assertEquals(bluetoothManager.getState(), State.SCO_CONNECTING);
+    simulateBluetoothScoConnectionConnected();
+    assertEquals(bluetoothManager.getState(), State.SCO_CONNECTED);
+    bluetoothManager.stopScoAudio();
+    simulateBluetoothScoConnectionDisconnected();
+    assertEquals(bluetoothManager.getState(), State.SCO_DISCONNECTING);
+    bluetoothManager.stop();
+    assertEquals(bluetoothManager.getState(), State.UNINITIALIZED);
+    verify(mockedAppRtcAudioManager, times(3)).updateAudioDeviceState();
+  }
+
+  /**
+   * Private helper methods.
+   */
+  private void simulateBluetoothServiceConnectedWithNoConnectedHeadset() {
+    mockedBluetoothDeviceList.clear();
+    when(mockedBluetoothHeadset.getConnectedDevices()).thenReturn(mockedBluetoothDeviceList);
+    bluetoothServiceListener.onServiceConnected(BluetoothProfile.HEADSET, mockedBluetoothHeadset);
+    // In real life, the AppRTC audio manager makes this call.
+    bluetoothManager.updateDevice();
+  }
+
+  private void simulateBluetoothServiceConnectedWithConnectedHeadset() {
+    mockedBluetoothDeviceList.clear();
+    mockedBluetoothDeviceList.add(mockedBluetoothDevice);
+    when(mockedBluetoothHeadset.getConnectedDevices()).thenReturn(mockedBluetoothDeviceList);
+    when(mockedBluetoothDevice.getName()).thenReturn(BLUETOOTH_TEST_DEVICE_NAME);
+    bluetoothServiceListener.onServiceConnected(BluetoothProfile.HEADSET, mockedBluetoothHeadset);
+    // In real life, the AppRTC audio manager makes this call.
+    bluetoothManager.updateDevice();
+  }
+
+  private void simulateBluetoothServiceDisconnected() {
+    bluetoothServiceListener.onServiceDisconnected(BluetoothProfile.HEADSET);
+  }
+
+  private void simulateBluetoothHeadsetConnected() {
+    Intent intent = new Intent();
+    intent.setAction(BluetoothHeadset.ACTION_CONNECTION_STATE_CHANGED);
+    intent.putExtra(BluetoothHeadset.EXTRA_STATE, BluetoothHeadset.STATE_CONNECTED);
+    bluetoothHeadsetStateReceiver.onReceive(context, intent);
+  }
+
+  private void simulateBluetoothScoConnectionConnected() {
+    Intent intent = new Intent();
+    intent.setAction(BluetoothHeadset.ACTION_AUDIO_STATE_CHANGED);
+    intent.putExtra(BluetoothHeadset.EXTRA_STATE, BluetoothHeadset.STATE_AUDIO_CONNECTED);
+    bluetoothHeadsetStateReceiver.onReceive(context, intent);
+  }
+
+  private void simulateBluetoothScoConnectionDisconnected() {
+    Intent intent = new Intent();
+    intent.setAction(BluetoothHeadset.ACTION_AUDIO_STATE_CHANGED);
+    intent.putExtra(BluetoothHeadset.EXTRA_STATE, BluetoothHeadset.STATE_AUDIO_DISCONNECTED);
+    bluetoothHeadsetStateReceiver.onReceive(context, intent);
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java b/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java
new file mode 100644
index 0000000000..2da8164ec7
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java
@@ -0,0 +1,155 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import static org.junit.Assert.assertFalse;
+import static org.junit.Assert.assertTrue;
+import static org.junit.Assert.fail;
+import static org.mockito.ArgumentMatchers.any;
+import static org.mockito.ArgumentMatchers.isNotNull;
+import static org.mockito.Mockito.mock;
+import static org.mockito.Mockito.timeout;
+import static org.mockito.Mockito.verify;
+import static org.mockito.Mockito.verifyNoMoreInteractions;
+
+import org.chromium.testing.local.LocalRobolectricTestRunner;
+import org.junit.Before;
+import org.junit.Test;
+import org.junit.runner.RunWith;
+import org.robolectric.annotation.Config;
+import org.robolectric.shadows.ShadowLog;
+import org.webrtc.IceCandidate;
+import org.webrtc.SessionDescription;
+
+/**
+ * Test for DirectRTCClient. Test is very simple and only tests the overall sanity of the class
+ * behaviour.
+ */
+@RunWith(LocalRobolectricTestRunner.class)
+@Config(manifest = Config.NONE)
+public class DirectRTCClientTest {
+  private static final String ROOM_URL = "";
+  private static final boolean LOOPBACK = false;
+
+  private static final String DUMMY_SDP_MID = "sdpMid";
+  private static final String DUMMY_SDP = "sdp";
+
+  public static final int SERVER_WAIT = 100;
+  public static final int NETWORK_TIMEOUT = 1000;
+
+  private DirectRTCClient client;
+  private DirectRTCClient server;
+
+  AppRTCClient.SignalingEvents clientEvents;
+  AppRTCClient.SignalingEvents serverEvents;
+
+  @Before
+  public void setUp() {
+    ShadowLog.stream = System.out;
+
+    clientEvents = mock(AppRTCClient.SignalingEvents.class);
+    serverEvents = mock(AppRTCClient.SignalingEvents.class);
+
+    client = new DirectRTCClient(clientEvents);
+    server = new DirectRTCClient(serverEvents);
+  }
+
+  @Test
+  public void testValidIpPattern() {
+    // Strings that should match the pattern.
+    // clang-format off
+    final String[] ipAddresses = new String[] {
+        "0.0.0.0",
+        "127.0.0.1",
+        "192.168.0.1",
+        "0.0.0.0:8888",
+        "127.0.0.1:8888",
+        "192.168.0.1:8888",
+        "::",
+        "::1",
+        "2001:0db8:85a3:0000:0000:8a2e:0370:7946",
+        "[::]",
+        "[::1]",
+        "[2001:0db8:85a3:0000:0000:8a2e:0370:7946]",
+        "[::]:8888",
+        "[::1]:8888",
+        "[2001:0db8:85a3:0000:0000:8a2e:0370:7946]:8888"
+    };
+    // clang-format on
+
+    for (String ip : ipAddresses) {
+      assertTrue(ip + " didn't match IP_PATTERN even though it should.",
+          DirectRTCClient.IP_PATTERN.matcher(ip).matches());
+    }
+  }
+
+  @Test
+  public void testInvalidIpPattern() {
+    // Strings that shouldn't match the pattern.
+    // clang-format off
+    final String[] invalidIpAddresses = new String[] {
+        "Hello, World!",
+        "aaaa",
+        "1111",
+        "[hello world]",
+        "hello:world"
+    };
+    // clang-format on
+
+    for (String invalidIp : invalidIpAddresses) {
+      assertFalse(invalidIp + " matched IP_PATTERN even though it shouldn't.",
+          DirectRTCClient.IP_PATTERN.matcher(invalidIp).matches());
+    }
+  }
+
+  // TODO(sakal): Replace isNotNull(class) with isNotNull() once Java 8 is used.
+  @SuppressWarnings("deprecation")
+  @Test
+  public void testDirectRTCClient() {
+    server.connectToRoom(new AppRTCClient.RoomConnectionParameters(ROOM_URL, "0.0.0.0", LOOPBACK));
+    try {
+      Thread.sleep(SERVER_WAIT);
+    } catch (InterruptedException e) {
+      fail(e.getMessage());
+    }
+    client.connectToRoom(
+        new AppRTCClient.RoomConnectionParameters(ROOM_URL, "127.0.0.1", LOOPBACK));
+    verify(serverEvents, timeout(NETWORK_TIMEOUT))
+        .onConnectedToRoom(any(AppRTCClient.SignalingParameters.class));
+
+    SessionDescription offerSdp = new SessionDescription(SessionDescription.Type.OFFER, DUMMY_SDP);
+    server.sendOfferSdp(offerSdp);
+    verify(clientEvents, timeout(NETWORK_TIMEOUT))
+        .onConnectedToRoom(any(AppRTCClient.SignalingParameters.class));
+
+    SessionDescription answerSdp =
+        new SessionDescription(SessionDescription.Type.ANSWER, DUMMY_SDP);
+    client.sendAnswerSdp(answerSdp);
+    verify(serverEvents, timeout(NETWORK_TIMEOUT))
+        .onRemoteDescription(isNotNull(SessionDescription.class));
+
+    IceCandidate candidate = new IceCandidate(DUMMY_SDP_MID, 0, DUMMY_SDP);
+    server.sendLocalIceCandidate(candidate);
+    verify(clientEvents, timeout(NETWORK_TIMEOUT))
+        .onRemoteIceCandidate(isNotNull(IceCandidate.class));
+
+    client.sendLocalIceCandidate(candidate);
+    verify(serverEvents, timeout(NETWORK_TIMEOUT))
+        .onRemoteIceCandidate(isNotNull(IceCandidate.class));
+
+    client.disconnectFromRoom();
+    verify(clientEvents, timeout(NETWORK_TIMEOUT)).onChannelClose();
+    verify(serverEvents, timeout(NETWORK_TIMEOUT)).onChannelClose();
+
+    verifyNoMoreInteractions(clientEvents);
+    verifyNoMoreInteractions(serverEvents);
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java b/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java
new file mode 100644
index 0000000000..b301d6317c
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java
@@ -0,0 +1,199 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc;
+
+import static org.junit.Assert.fail;
+import static org.mockito.Mockito.timeout;
+import static org.mockito.Mockito.verify;
+import static org.mockito.Mockito.verifyNoMoreInteractions;
+
+import org.chromium.testing.local.LocalRobolectricTestRunner;
+import org.junit.After;
+import org.junit.Before;
+import org.junit.Test;
+import org.junit.runner.RunWith;
+import org.mockito.Mock;
+import org.mockito.MockitoAnnotations;
+import org.robolectric.annotation.Config;
+import org.robolectric.shadows.ShadowLog;
+
+import java.util.concurrent.ExecutorService;
+import java.util.concurrent.Executors;
+import java.util.concurrent.TimeUnit;
+
+@RunWith(LocalRobolectricTestRunner.class)
+@Config(manifest = Config.NONE)
+public class TCPChannelClientTest {
+  private static final int PORT = 8888;
+  /**
+   * How long we wait before trying to connect to the server. Note: was
+   * previously only 10, which was too short (tests were flaky).
+   */
+  private static final int SERVER_WAIT = 300;
+  private static final int CONNECT_TIMEOUT = 1000;
+  private static final int SEND_TIMEOUT = 1000;
+  private static final int DISCONNECT_TIMEOUT = 1000;
+  private static final int TERMINATION_TIMEOUT = 1000;
+  private static final String TEST_MESSAGE_SERVER = "Hello, Server!";
+  private static final String TEST_MESSAGE_CLIENT = "Hello, Client!";
+
+  @Mock TCPChannelClient.TCPChannelEvents serverEvents;
+  @Mock TCPChannelClient.TCPChannelEvents clientEvents;
+
+  private ExecutorService executor;
+  private TCPChannelClient server;
+  private TCPChannelClient client;
+
+  @Before
+  public void setUp() {
+    ShadowLog.stream = System.out;
+
+    MockitoAnnotations.initMocks(this);
+
+    executor = Executors.newSingleThreadExecutor();
+  }
+
+  @After
+  public void tearDown() {
+    verifyNoMoreEvents();
+
+    executeAndWait(new Runnable() {
+      @Override
+      public void run() {
+        client.disconnect();
+        server.disconnect();
+      }
+    });
+
+    // Stop the executor thread
+    executor.shutdown();
+    try {
+      executor.awaitTermination(TERMINATION_TIMEOUT, TimeUnit.MILLISECONDS);
+    } catch (InterruptedException e) {
+      fail(e.getMessage());
+    }
+  }
+
+  @Test
+  public void testConnectIPv4() {
+    setUpIPv4Server();
+    try {
+      Thread.sleep(SERVER_WAIT);
+    } catch (InterruptedException e) {
+      fail(e.getMessage());
+    }
+    setUpIPv4Client();
+
+    verify(serverEvents, timeout(CONNECT_TIMEOUT)).onTCPConnected(true);
+    verify(clientEvents, timeout(CONNECT_TIMEOUT)).onTCPConnected(false);
+  }
+
+  @Test
+  public void testConnectIPv6() {
+    setUpIPv6Server();
+    try {
+      Thread.sleep(SERVER_WAIT);
+    } catch (InterruptedException e) {
+      fail(e.getMessage());
+    }
+    setUpIPv6Client();
+
+    verify(serverEvents, timeout(CONNECT_TIMEOUT)).onTCPConnected(true);
+    verify(clientEvents, timeout(CONNECT_TIMEOUT)).onTCPConnected(false);
+  }
+
+  @Test
+  public void testSendData() {
+    testConnectIPv4();
+
+    executeAndWait(new Runnable() {
+      @Override
+      public void run() {
+        client.send(TEST_MESSAGE_SERVER);
+        server.send(TEST_MESSAGE_CLIENT);
+      }
+    });
+
+    verify(serverEvents, timeout(SEND_TIMEOUT)).onTCPMessage(TEST_MESSAGE_SERVER);
+    verify(clientEvents, timeout(SEND_TIMEOUT)).onTCPMessage(TEST_MESSAGE_CLIENT);
+  }
+
+  @Test
+  public void testDisconnectServer() {
+    testConnectIPv4();
+    executeAndWait(new Runnable() {
+      @Override
+      public void run() {
+        server.disconnect();
+      }
+    });
+
+    verify(serverEvents, timeout(DISCONNECT_TIMEOUT)).onTCPClose();
+    verify(clientEvents, timeout(DISCONNECT_TIMEOUT)).onTCPClose();
+  }
+
+  @Test
+  public void testDisconnectClient() {
+    testConnectIPv4();
+    executeAndWait(new Runnable() {
+      @Override
+      public void run() {
+        client.disconnect();
+      }
+    });
+
+    verify(serverEvents, timeout(DISCONNECT_TIMEOUT)).onTCPClose();
+    verify(clientEvents, timeout(DISCONNECT_TIMEOUT)).onTCPClose();
+  }
+
+  private void setUpIPv4Server() {
+    setUpServer("0.0.0.0", PORT);
+  }
+
+  private void setUpIPv4Client() {
+    setUpClient("127.0.0.1", PORT);
+  }
+
+  private void setUpIPv6Server() {
+    setUpServer("::", PORT);
+  }
+
+  private void setUpIPv6Client() {
+    setUpClient("::1", PORT);
+  }
+
+  private void setUpServer(String ip, int port) {
+    server = new TCPChannelClient(executor, serverEvents, ip, port);
+  }
+
+  private void setUpClient(String ip, int port) {
+    client = new TCPChannelClient(executor, clientEvents, ip, port);
+  }
+
+  /**
+   * Verifies no more server or client events have been issued
+   */
+  private void verifyNoMoreEvents() {
+    verifyNoMoreInteractions(serverEvents);
+    verifyNoMoreInteractions(clientEvents);
+  }
+
+  /**
+   * Queues runnable to be run and waits for it to be executed by the executor thread
+   */
+  public void executeAndWait(Runnable runnable) {
+    try {
+      executor.submit(runnable).get();
+    } catch (Exception e) {
+      fail(e.getMessage());
+    }
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidnativeapi/AndroidManifest.xml b/third_party/libwebrtc/examples/androidnativeapi/AndroidManifest.xml
new file mode 100644
index 0000000000..27c7d9dd35
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidnativeapi/AndroidManifest.xml
@@ -0,0 +1,23 @@
+<?xml version="1.0" encoding="utf-8"?>
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+  package="org.webrtc.examples.androidnativeapi">
+
+
+  <uses-permission android:name="android.permission.INTERNET" />
+  <uses-permission android:name="android.permission.CAMERA" />
+
+  <application
+    android:allowBackup="true"
+    android:label="@string/app_name"
+    android:supportsRtl="true">
+    <activity android:name=".MainActivity"
+      android:exported="true">
+      <intent-filter>
+        <action android:name="android.intent.action.MAIN" />
+
+        <category android:name="android.intent.category.LAUNCHER" />
+      </intent-filter>
+    </activity>
+  </application>
+
+</manifest>
diff --git a/third_party/libwebrtc/examples/androidnativeapi/BUILD.gn b/third_party/libwebrtc/examples/androidnativeapi/BUILD.gn
new file mode 100644
index 0000000000..e4c48a2b36
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidnativeapi/BUILD.gn
@@ -0,0 +1,80 @@
+import("//webrtc.gni")
+
+if (is_android) {
+  rtc_android_apk("androidnativeapi") {
+    testonly = true
+    apk_name = "androidnativeapi"
+    android_manifest = "AndroidManifest.xml"
+    min_sdk_version = 21
+    target_sdk_version = 31
+
+    sources = [
+      "java/org/webrtc/examples/androidnativeapi/CallClient.java",
+      "java/org/webrtc/examples/androidnativeapi/MainActivity.java",
+    ]
+
+    deps = [
+      ":resources",
+      "//modules/audio_device:audio_device_java",
+      "//rtc_base:base_java",
+      "//sdk/android:camera_java",
+      "//sdk/android:surfaceviewrenderer_java",
+      "//sdk/android:video_api_java",
+      "//sdk/android:video_java",
+      "//third_party/androidx:androidx_annotation_annotation_java",
+    ]
+
+    shared_libraries = [ ":examples_androidnativeapi_jni" ]
+  }
+
+  generate_jni("generated_jni") {
+    testonly = true
+    sources = [ "java/org/webrtc/examples/androidnativeapi/CallClient.java" ]
+    namespace = "webrtc_examples"
+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
+  }
+
+  rtc_shared_library("examples_androidnativeapi_jni") {
+    testonly = true
+    sources = [
+      "jni/android_call_client.cc",
+      "jni/android_call_client.h",
+      "jni/onload.cc",
+    ]
+
+    suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
+    configs += [ "//build/config/android:hide_all_but_jni" ]
+
+    deps = [
+      ":generated_jni",
+      "../../api:scoped_refptr",
+      "../../api:sequence_checker",
+      "../../rtc_base:ssl",
+      "../../rtc_base/synchronization:mutex",
+      "//api:libjingle_peerconnection_api",
+      "//api/rtc_event_log:rtc_event_log_factory",
+      "//api/task_queue:default_task_queue_factory",
+      "//media:rtc_audio_video",
+      "//media:rtc_internal_video_codecs",
+      "//media:rtc_media_engine_defaults",
+      "//modules/utility",
+      "//pc:libjingle_peerconnection",
+      "//sdk/android:native_api_base",
+      "//sdk/android:native_api_jni",
+      "//sdk/android:native_api_video",
+    ]
+  }
+
+  android_resources("resources") {
+    testonly = true
+    custom_package = "org.webrtc.examples.androidnativeapi"
+    sources = [
+      "res/layout/activity_main.xml",
+      "res/values/strings.xml",
+    ]
+
+    # Needed for Bazel converter.
+    resource_dirs = [ "res" ]
+    assert(resource_dirs != [])  # Mark as used.
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidnativeapi/DEPS b/third_party/libwebrtc/examples/androidnativeapi/DEPS
new file mode 100644
index 0000000000..5dbfcf1bb8
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidnativeapi/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+  "+logging/rtc_event_log/rtc_event_log_factory.h",
+  "+modules/utility/include",
+  "+sdk/android/native_api",
+]
diff --git a/third_party/libwebrtc/examples/androidnativeapi/OWNERS b/third_party/libwebrtc/examples/androidnativeapi/OWNERS
new file mode 100644
index 0000000000..cf092a316a
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidnativeapi/OWNERS
@@ -0,0 +1 @@
+xalep@webrtc.org
diff --git a/third_party/libwebrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/CallClient.java b/third_party/libwebrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/CallClient.java
new file mode 100644
index 0000000000..7369a1286d
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/CallClient.java
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.examples.androidnativeapi;
+
+import android.content.Context;
+import android.os.Handler;
+import android.os.HandlerThread;
+import org.webrtc.CapturerObserver;
+import org.webrtc.SurfaceTextureHelper;
+import org.webrtc.VideoCapturer;
+import org.webrtc.VideoSink;
+
+public class CallClient {
+  private static final String TAG = "CallClient";
+  private static final int CAPTURE_WIDTH = 640;
+  private static final int CAPTURE_HEIGHT = 480;
+  private static final int CAPTURE_FPS = 30;
+
+  private final Context applicationContext;
+  private final HandlerThread thread;
+  private final Handler handler;
+
+  private long nativeClient;
+  private SurfaceTextureHelper surfaceTextureHelper;
+  private VideoCapturer videoCapturer;
+
+  public CallClient(Context applicationContext) {
+    this.applicationContext = applicationContext;
+    thread = new HandlerThread(TAG + "Thread");
+    thread.start();
+    handler = new Handler(thread.getLooper());
+    handler.post(() -> { nativeClient = nativeCreateClient(); });
+  }
+
+  public void call(VideoSink localSink, VideoSink remoteSink, VideoCapturer videoCapturer,
+      SurfaceTextureHelper videoCapturerSurfaceTextureHelper) {
+    handler.post(() -> {
+      nativeCall(nativeClient, localSink, remoteSink);
+      videoCapturer.initialize(videoCapturerSurfaceTextureHelper, applicationContext,
+          nativeGetJavaVideoCapturerObserver(nativeClient));
+      videoCapturer.startCapture(CAPTURE_WIDTH, CAPTURE_HEIGHT, CAPTURE_FPS);
+    });
+  }
+
+  public void hangup() {
+    handler.post(() -> { nativeHangup(nativeClient); });
+  }
+
+  public void close() {
+    handler.post(() -> {
+      nativeDelete(nativeClient);
+      nativeClient = 0;
+    });
+    thread.quitSafely();
+  }
+
+  private static native long nativeCreateClient();
+  private static native void nativeCall(
+      long nativeAndroidCallClient, VideoSink localSink, VideoSink remoteSink);
+  private static native void nativeHangup(long nativeAndroidCallClient);
+  private static native void nativeDelete(long nativeAndroidCallClient);
+  private static native CapturerObserver nativeGetJavaVideoCapturerObserver(
+      long nativeAndroidCallClient);
+}
diff --git a/third_party/libwebrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/MainActivity.java b/third_party/libwebrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/MainActivity.java
new file mode 100644
index 0000000000..72fc0a686d
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/MainActivity.java
@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.examples.androidnativeapi;
+
+import android.app.Activity;
+import android.content.Context;
+import android.os.Bundle;
+import android.widget.Button;
+import androidx.annotation.Nullable;
+import org.webrtc.Camera1Enumerator;
+import org.webrtc.Camera2Enumerator;
+import org.webrtc.CameraEnumerator;
+import org.webrtc.ContextUtils;
+import org.webrtc.EglBase;
+import org.webrtc.GlRectDrawer;
+import org.webrtc.SurfaceTextureHelper;
+import org.webrtc.SurfaceViewRenderer;
+import org.webrtc.VideoCapturer;
+
+public class MainActivity extends Activity {
+  private @Nullable CallClient callClient;
+  private @Nullable EglBase eglBase;
+  private @Nullable SurfaceViewRenderer localRenderer;
+  private @Nullable SurfaceViewRenderer remoteRenderer;
+  private @Nullable SurfaceTextureHelper videoCapturerSurfaceTextureHelper;
+  private @Nullable VideoCapturer videoCapturer;
+
+  @Override
+  protected void onCreate(Bundle savedInstance) {
+    ContextUtils.initialize(getApplicationContext());
+
+    super.onCreate(savedInstance);
+    setContentView(R.layout.activity_main);
+
+    System.loadLibrary("examples_androidnativeapi_jni");
+    callClient = new CallClient(getApplicationContext());
+
+    Button callButton = (Button) findViewById(R.id.call_button);
+    callButton.setOnClickListener((view) -> {
+      if (videoCapturer == null) {
+        videoCapturer = createVideoCapturer(getApplicationContext());
+      }
+      callClient.call(
+          localRenderer, remoteRenderer, videoCapturer, videoCapturerSurfaceTextureHelper);
+    });
+
+    Button hangupButton = (Button) findViewById(R.id.hangup_button);
+    hangupButton.setOnClickListener((view) -> { hangup(); });
+  }
+
+  @Override
+  protected void onStart() {
+    super.onStart();
+
+    eglBase = EglBase.create(null /* sharedContext */, EglBase.CONFIG_PLAIN);
+    localRenderer = (SurfaceViewRenderer) findViewById(R.id.local_renderer);
+    remoteRenderer = (SurfaceViewRenderer) findViewById(R.id.remote_renderer);
+
+    localRenderer.init(eglBase.getEglBaseContext(), null /* rendererEvents */, EglBase.CONFIG_PLAIN,
+        new GlRectDrawer());
+    remoteRenderer.init(eglBase.getEglBaseContext(), null /* rendererEvents */,
+        EglBase.CONFIG_PLAIN, new GlRectDrawer());
+
+    videoCapturerSurfaceTextureHelper =
+        SurfaceTextureHelper.create("VideoCapturerThread", eglBase.getEglBaseContext());
+  }
+
+  @Override
+  protected void onStop() {
+    hangup();
+
+    localRenderer.release();
+    remoteRenderer.release();
+    videoCapturerSurfaceTextureHelper.dispose();
+    eglBase.release();
+
+    localRenderer = null;
+    remoteRenderer = null;
+    videoCapturerSurfaceTextureHelper = null;
+    eglBase = null;
+
+    super.onStop();
+  }
+
+  @Override
+  protected void onDestroy() {
+    callClient.close();
+    callClient = null;
+
+    super.onDestroy();
+  }
+
+  private void hangup() {
+    if (videoCapturer != null) {
+      try {
+        videoCapturer.stopCapture();
+      } catch (InterruptedException e) {
+        throw new RuntimeException(e);
+      }
+      videoCapturer.dispose();
+      videoCapturer = null;
+    }
+    callClient.hangup();
+  }
+
+  private static VideoCapturer createVideoCapturer(Context context) {
+    CameraEnumerator enumerator = Camera2Enumerator.isSupported(context)
+        ? new Camera2Enumerator(context)
+        : new Camera1Enumerator();
+    return enumerator.createCapturer(enumerator.getDeviceNames()[0], null /* eventsHandler */);
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.cc b/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.cc
new file mode 100644
index 0000000000..7da56e6e60
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.cc
@@ -0,0 +1,294 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/androidnativeapi/jni/android_call_client.h"
+
+#include <utility>
+
+#include <memory>
+
+#include "api/peer_connection_interface.h"
+#include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "examples/androidnativeapi/generated_jni/CallClient_jni.h"
+#include "media/engine/internal_decoder_factory.h"
+#include "media/engine/internal_encoder_factory.h"
+#include "media/engine/webrtc_media_engine.h"
+#include "media/engine/webrtc_media_engine_defaults.h"
+#include "sdk/android/native_api/jni/java_types.h"
+#include "sdk/android/native_api/video/wrapper.h"
+
+namespace webrtc_examples {
+
+class AndroidCallClient::PCObserver : public webrtc::PeerConnectionObserver {
+ public:
+  explicit PCObserver(AndroidCallClient* client);
+
+  void OnSignalingChange(
+      webrtc::PeerConnectionInterface::SignalingState new_state) override;
+  void OnDataChannel(
+      rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;
+  void OnRenegotiationNeeded() override;
+  void OnIceConnectionChange(
+      webrtc::PeerConnectionInterface::IceConnectionState new_state) override;
+  void OnIceGatheringChange(
+      webrtc::PeerConnectionInterface::IceGatheringState new_state) override;
+  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
+
+ private:
+  AndroidCallClient* const client_;
+};
+
+namespace {
+
+class CreateOfferObserver : public webrtc::CreateSessionDescriptionObserver {
+ public:
+  explicit CreateOfferObserver(
+      rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc);
+
+  void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
+  void OnFailure(webrtc::RTCError error) override;
+
+ private:
+  const rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_;
+};
+
+class SetRemoteSessionDescriptionObserver
+    : public webrtc::SetRemoteDescriptionObserverInterface {
+ public:
+  void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override;
+};
+
+class SetLocalSessionDescriptionObserver
+    : public webrtc::SetSessionDescriptionObserver {
+ public:
+  void OnSuccess() override;
+  void OnFailure(webrtc::RTCError error) override;
+};
+
+}  // namespace
+
+AndroidCallClient::AndroidCallClient()
+    : call_started_(false), pc_observer_(std::make_unique<PCObserver>(this)) {
+  thread_checker_.Detach();
+  CreatePeerConnectionFactory();
+}
+
+AndroidCallClient::~AndroidCallClient() = default;
+
+void AndroidCallClient::Call(JNIEnv* env,
+                             const webrtc::JavaRef<jobject>& local_sink,
+                             const webrtc::JavaRef<jobject>& remote_sink) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  webrtc::MutexLock lock(&pc_mutex_);
+  if (call_started_) {
+    RTC_LOG(LS_WARNING) << "Call already started.";
+    return;
+  }
+  call_started_ = true;
+
+  local_sink_ = webrtc::JavaToNativeVideoSink(env, local_sink.obj());
+  remote_sink_ = webrtc::JavaToNativeVideoSink(env, remote_sink.obj());
+
+  video_source_ = webrtc::CreateJavaVideoSource(env, signaling_thread_.get(),
+                                                /* is_screencast= */ false,
+                                                /* align_timestamps= */ true);
+
+  CreatePeerConnection();
+  Connect();
+}
+
+void AndroidCallClient::Hangup(JNIEnv* env) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  call_started_ = false;
+
+  {
+    webrtc::MutexLock lock(&pc_mutex_);
+    if (pc_ != nullptr) {
+      pc_->Close();
+      pc_ = nullptr;
+    }
+  }
+
+  local_sink_ = nullptr;
+  remote_sink_ = nullptr;
+  video_source_ = nullptr;
+}
+
+void AndroidCallClient::Delete(JNIEnv* env) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  delete this;
+}
+
+webrtc::ScopedJavaLocalRef<jobject>
+AndroidCallClient::GetJavaVideoCapturerObserver(JNIEnv* env) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  return video_source_->GetJavaVideoCapturerObserver(env);
+}
+
+void AndroidCallClient::CreatePeerConnectionFactory() {
+  network_thread_ = rtc::Thread::CreateWithSocketServer();
+  network_thread_->SetName("network_thread", nullptr);
+  RTC_CHECK(network_thread_->Start()) << "Failed to start thread";
+
+  worker_thread_ = rtc::Thread::Create();
+  worker_thread_->SetName("worker_thread", nullptr);
+  RTC_CHECK(worker_thread_->Start()) << "Failed to start thread";
+
+  signaling_thread_ = rtc::Thread::Create();
+  signaling_thread_->SetName("signaling_thread", nullptr);
+  RTC_CHECK(signaling_thread_->Start()) << "Failed to start thread";
+
+  webrtc::PeerConnectionFactoryDependencies pcf_deps;
+  pcf_deps.network_thread = network_thread_.get();
+  pcf_deps.worker_thread = worker_thread_.get();
+  pcf_deps.signaling_thread = signaling_thread_.get();
+  pcf_deps.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
+  pcf_deps.call_factory = webrtc::CreateCallFactory();
+  pcf_deps.event_log_factory = std::make_unique<webrtc::RtcEventLogFactory>(
+      pcf_deps.task_queue_factory.get());
+
+  cricket::MediaEngineDependencies media_deps;
+  media_deps.task_queue_factory = pcf_deps.task_queue_factory.get();
+  media_deps.video_encoder_factory =
+      std::make_unique<webrtc::InternalEncoderFactory>();
+  media_deps.video_decoder_factory =
+      std::make_unique<webrtc::InternalDecoderFactory>();
+  webrtc::SetMediaEngineDefaults(&media_deps);
+  pcf_deps.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
+  RTC_LOG(LS_INFO) << "Media engine created: " << pcf_deps.media_engine.get();
+
+  pcf_ = CreateModularPeerConnectionFactory(std::move(pcf_deps));
+  RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_.get();
+}
+
+void AndroidCallClient::CreatePeerConnection() {
+  webrtc::MutexLock lock(&pc_mutex_);
+  webrtc::PeerConnectionInterface::RTCConfiguration config;
+  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+  // Encryption has to be disabled for loopback to work.
+  webrtc::PeerConnectionFactoryInterface::Options options;
+  options.disable_encryption = true;
+  pcf_->SetOptions(options);
+  webrtc::PeerConnectionDependencies deps(pc_observer_.get());
+  pc_ = pcf_->CreatePeerConnectionOrError(config, std::move(deps)).MoveValue();
+
+  RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_.get();
+
+  rtc::scoped_refptr<webrtc::VideoTrackInterface> local_video_track(
+      pcf_->CreateVideoTrack("video", video_source_.get()));
+  local_video_track->AddOrUpdateSink(local_sink_.get(), rtc::VideoSinkWants());
+  pc_->AddTransceiver(local_video_track);
+  RTC_LOG(LS_INFO) << "Local video sink set up: " << local_video_track.get();
+
+  for (const rtc::scoped_refptr<webrtc::RtpTransceiverInterface>& tranceiver :
+       pc_->GetTransceivers()) {
+    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track =
+        tranceiver->receiver()->track();
+    if (track &&
+        track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) {
+      static_cast<webrtc::VideoTrackInterface*>(track.get())
+          ->AddOrUpdateSink(remote_sink_.get(), rtc::VideoSinkWants());
+      RTC_LOG(LS_INFO) << "Remote video sink set up: " << track.get();
+      break;
+    }
+  }
+}
+
+void AndroidCallClient::Connect() {
+  webrtc::MutexLock lock(&pc_mutex_);
+  pc_->CreateOffer(rtc::make_ref_counted<CreateOfferObserver>(pc_).get(),
+                   webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
+}
+
+AndroidCallClient::PCObserver::PCObserver(AndroidCallClient* client)
+    : client_(client) {}
+
+void AndroidCallClient::PCObserver::OnSignalingChange(
+    webrtc::PeerConnectionInterface::SignalingState new_state) {
+  RTC_LOG(LS_INFO) << "OnSignalingChange: " << new_state;
+}
+
+void AndroidCallClient::PCObserver::OnDataChannel(
+    rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) {
+  RTC_LOG(LS_INFO) << "OnDataChannel";
+}
+
+void AndroidCallClient::PCObserver::OnRenegotiationNeeded() {
+  RTC_LOG(LS_INFO) << "OnRenegotiationNeeded";
+}
+
+void AndroidCallClient::PCObserver::OnIceConnectionChange(
+    webrtc::PeerConnectionInterface::IceConnectionState new_state) {
+  RTC_LOG(LS_INFO) << "OnIceConnectionChange: " << new_state;
+}
+
+void AndroidCallClient::PCObserver::OnIceGatheringChange(
+    webrtc::PeerConnectionInterface::IceGatheringState new_state) {
+  RTC_LOG(LS_INFO) << "OnIceGatheringChange: " << new_state;
+}
+
+void AndroidCallClient::PCObserver::OnIceCandidate(
+    const webrtc::IceCandidateInterface* candidate) {
+  RTC_LOG(LS_INFO) << "OnIceCandidate: " << candidate->server_url();
+  webrtc::MutexLock lock(&client_->pc_mutex_);
+  RTC_DCHECK(client_->pc_ != nullptr);
+  client_->pc_->AddIceCandidate(candidate);
+}
+
+CreateOfferObserver::CreateOfferObserver(
+    rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc)
+    : pc_(pc) {}
+
+void CreateOfferObserver::OnSuccess(webrtc::SessionDescriptionInterface* desc) {
+  std::string sdp;
+  desc->ToString(&sdp);
+  RTC_LOG(LS_INFO) << "Created offer: " << sdp;
+
+  // Ownership of desc was transferred to us, now we transfer it forward.
+  pc_->SetLocalDescription(
+      rtc::make_ref_counted<SetLocalSessionDescriptionObserver>().get(), desc);
+
+  // Generate a fake answer.
+  std::unique_ptr<webrtc::SessionDescriptionInterface> answer(
+      webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp));
+  pc_->SetRemoteDescription(
+      std::move(answer),
+      rtc::make_ref_counted<SetRemoteSessionDescriptionObserver>());
+}
+
+void CreateOfferObserver::OnFailure(webrtc::RTCError error) {
+  RTC_LOG(LS_INFO) << "Failed to create offer: " << ToString(error.type())
+                   << ": " << error.message();
+}
+
+void SetRemoteSessionDescriptionObserver::OnSetRemoteDescriptionComplete(
+    webrtc::RTCError error) {
+  RTC_LOG(LS_INFO) << "Set remote description: " << error.message();
+}
+
+void SetLocalSessionDescriptionObserver::OnSuccess() {
+  RTC_LOG(LS_INFO) << "Set local description success!";
+}
+
+void SetLocalSessionDescriptionObserver::OnFailure(webrtc::RTCError error) {
+  RTC_LOG(LS_INFO) << "Set local description failure: "
+                   << ToString(error.type()) << ": " << error.message();
+}
+
+static jlong JNI_CallClient_CreateClient(JNIEnv* env) {
+  return webrtc::NativeToJavaPointer(new webrtc_examples::AndroidCallClient());
+}
+
+}  // namespace webrtc_examples
diff --git a/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.h b/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.h
new file mode 100644
index 0000000000..c9153d09bd
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidnativeapi/jni/android_call_client.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_ANDROIDNATIVEAPI_JNI_ANDROID_CALL_CLIENT_H_
+#define EXAMPLES_ANDROIDNATIVEAPI_JNI_ANDROID_CALL_CLIENT_H_
+
+#include <jni.h>
+
+#include <memory>
+#include <string>
+
+#include "api/peer_connection_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "sdk/android/native_api/jni/scoped_java_ref.h"
+#include "sdk/android/native_api/video/video_source.h"
+
+namespace webrtc_examples {
+
+class AndroidCallClient {
+ public:
+  AndroidCallClient();
+  ~AndroidCallClient();
+
+  void Call(JNIEnv* env,
+            const webrtc::JavaRef<jobject>& local_sink,
+            const webrtc::JavaRef<jobject>& remote_sink);
+  void Hangup(JNIEnv* env);
+  // A helper method for Java code to delete this object. Calls delete this.
+  void Delete(JNIEnv* env);
+
+  webrtc::ScopedJavaLocalRef<jobject> GetJavaVideoCapturerObserver(JNIEnv* env);
+
+ private:
+  class PCObserver;
+
+  void CreatePeerConnectionFactory() RTC_RUN_ON(thread_checker_);
+  void CreatePeerConnection() RTC_RUN_ON(thread_checker_);
+  void Connect() RTC_RUN_ON(thread_checker_);
+
+  webrtc::SequenceChecker thread_checker_;
+
+  bool call_started_ RTC_GUARDED_BY(thread_checker_);
+
+  const std::unique_ptr<PCObserver> pc_observer_;
+
+  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf_
+      RTC_GUARDED_BY(thread_checker_);
+  std::unique_ptr<rtc::Thread> network_thread_ RTC_GUARDED_BY(thread_checker_);
+  std::unique_ptr<rtc::Thread> worker_thread_ RTC_GUARDED_BY(thread_checker_);
+  std::unique_ptr<rtc::Thread> signaling_thread_
+      RTC_GUARDED_BY(thread_checker_);
+
+  std::unique_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> local_sink_
+      RTC_GUARDED_BY(thread_checker_);
+  std::unique_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> remote_sink_
+      RTC_GUARDED_BY(thread_checker_);
+  rtc::scoped_refptr<webrtc::JavaVideoTrackSourceInterface> video_source_
+      RTC_GUARDED_BY(thread_checker_);
+
+  webrtc::Mutex pc_mutex_;
+  rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_
+      RTC_GUARDED_BY(pc_mutex_);
+};
+
+}  // namespace webrtc_examples
+
+#endif  // EXAMPLES_ANDROIDNATIVEAPI_JNI_ANDROID_CALL_CLIENT_H_
diff --git a/third_party/libwebrtc/examples/androidnativeapi/jni/onload.cc b/third_party/libwebrtc/examples/androidnativeapi/jni/onload.cc
new file mode 100644
index 0000000000..6ea5275d2a
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidnativeapi/jni/onload.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <jni.h>
+
+#include "modules/utility/include/jvm_android.h"
+#include "rtc_base/ssl_adapter.h"
+#include "sdk/android/native_api/base/init.h"
+
+namespace webrtc_examples {
+
+extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM* jvm, void* reserved) {
+  webrtc::InitAndroid(jvm);
+  webrtc::JVM::Initialize(jvm);
+  RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
+  return JNI_VERSION_1_6;
+}
+
+extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM* jvm, void* reserved) {
+  RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()";
+}
+
+}  // namespace webrtc_examples
diff --git a/third_party/libwebrtc/examples/androidnativeapi/res/layout/activity_main.xml b/third_party/libwebrtc/examples/androidnativeapi/res/layout/activity_main.xml
new file mode 100644
index 0000000000..ac8037320f
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidnativeapi/res/layout/activity_main.xml
@@ -0,0 +1,52 @@
+<?xml version="1.0" encoding="utf-8"?>
+<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
+  xmlns:app="http://schemas.android.com/apk/res-auto"
+  xmlns:tools="http://schemas.android.com/tools"
+  android:orientation="vertical"
+  android:layout_width="match_parent"
+  android:layout_height="match_parent"
+  android:padding="8dp"
+  tools:context="org.webrtc.examples.androidnativeapi.MainActivity">
+
+  <org.webrtc.SurfaceViewRenderer
+    android:id="@+id/local_renderer"
+    android:layout_width="match_parent"
+    android:layout_height="0dp"
+    android:layout_weight="1"
+    android:layout_margin="8dp"/>
+
+  <org.webrtc.SurfaceViewRenderer
+    android:id="@+id/remote_renderer"
+    android:layout_width="match_parent"
+    android:layout_height="0dp"
+    android:layout_weight="1"
+    android:layout_margin="8dp"/>
+
+
+  <LinearLayout
+    android:orientation="horizontal"
+    android:layout_width="match_parent"
+    android:layout_height="48dp"
+    style="?android:attr/buttonBarStyle">
+
+    <Button
+      android:id="@+id/call_button"
+      android:text="@string/call_button"
+      style="?android:attr/buttonBarButtonStyle"
+      android:layout_width="0dp"
+      android:layout_height="48dp"
+      android:layout_weight="1"
+      android:layout_margin="8dp"/>
+
+    <Button
+      android:id="@+id/hangup_button"
+      android:text="@string/hangup_button"
+      style="?android:attr/buttonBarButtonStyle"
+      android:layout_width="0dp"
+      android:layout_height="48dp"
+      android:layout_weight="1"
+      android:layout_margin="8dp"/>
+
+  </LinearLayout>
+
+</LinearLayout>
diff --git a/third_party/libwebrtc/examples/androidnativeapi/res/values/strings.xml b/third_party/libwebrtc/examples/androidnativeapi/res/values/strings.xml
new file mode 100644
index 0000000000..a00920c92b
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidnativeapi/res/values/strings.xml
@@ -0,0 +1,5 @@
+<resources>
+  <string name="app_name">androidnativeapi</string>
+  <string name="call_button">Call</string>
+  <string name="hangup_button">Hangup</string>
+</resources>
diff --git a/third_party/libwebrtc/examples/androidtests/AndroidManifest.xml b/third_party/libwebrtc/examples/androidtests/AndroidManifest.xml
new file mode 100644
index 0000000000..38ed3e3b9a
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidtests/AndroidManifest.xml
@@ -0,0 +1,26 @@
+<!--
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+-->
+
+<manifest
+      xmlns:android="http://schemas.android.com/apk/res/android"
+      xmlns:tools="http://schemas.android.com/tools"
+      package="org.appspot.apprtc.test">
+
+    <uses-permission android:name="android.permission.RUN_INSTRUMENTATION" />
+
+    <application>
+        <uses-library android:name="android.test.runner" />
+    </application>
+
+    <instrumentation android:name="android.support.test.runner.AndroidJUnitRunner"
+        tools:ignore="MissingPrefix"
+        android:targetPackage="org.appspot.apprtc"
+        android:label="Tests for AppRTCMobile"/>
+</manifest>
diff --git a/third_party/libwebrtc/examples/androidtests/OWNERS b/third_party/libwebrtc/examples/androidtests/OWNERS
new file mode 100644
index 0000000000..cf092a316a
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidtests/OWNERS
@@ -0,0 +1 @@
+xalep@webrtc.org
diff --git a/third_party/libwebrtc/examples/androidtests/README b/third_party/libwebrtc/examples/androidtests/README
new file mode 100644
index 0000000000..0701b0e896
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidtests/README
@@ -0,0 +1,14 @@
+This directory contains an example unit test for Android AppRTCMobile.
+
+Example of building & using the app:
+
+- Build Android AppRTCMobile and AppRTCMobile unit test:
+cd <path/to/webrtc>/src
+ninja -C out/Debug AppRTCMobile_test_apk
+
+- Install AppRTCMobile and AppRTCMobileTest:
+adb install -r out/Debug/apks/AppRTCMobile.apk
+adb install -r out/Debug/apks/AppRTCMobileTest.apk
+
+- Run unit tests:
+adb shell am instrument -w org.appspot.apprtc.test/android.test.InstrumentationTestRunner
diff --git a/third_party/libwebrtc/examples/androidtests/ant.properties b/third_party/libwebrtc/examples/androidtests/ant.properties
new file mode 100644
index 0000000000..ec7d042885
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidtests/ant.properties
@@ -0,0 +1,18 @@
+# This file is used to override default values used by the Ant build system.
+#
+# This file must be checked into Version Control Systems, as it is
+# integral to the build system of your project.
+
+# This file is only used by the Ant script.
+
+# You can use this to override default values such as
+#  'source.dir' for the location of your java source folder and
+#  'out.dir' for the location of your output folder.
+
+# You can also use it define how the release builds are signed by declaring
+# the following properties:
+#  'key.store' for the location of your keystore and
+#  'key.alias' for the name of the key to use.
+# The password will be asked during the build when you use the 'release' target.
+
+tested.project.dir=../android
diff --git a/third_party/libwebrtc/examples/androidtests/build.xml b/third_party/libwebrtc/examples/androidtests/build.xml
new file mode 100644
index 0000000000..95847b74c2
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidtests/build.xml
@@ -0,0 +1,92 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<project name="AppRTCMobileTest" default="help">
+
+    <!-- The local.properties file is created and updated by the 'android' tool.
+         It contains the path to the SDK. It should *NOT* be checked into
+         Version Control Systems. -->
+    <property file="local.properties" />
+
+    <!-- The ant.properties file can be created by you. It is only edited by the
+         'android' tool to add properties to it.
+         This is the place to change some Ant specific build properties.
+         Here are some properties you may want to change/update:
+
+         source.dir
+             The name of the source directory. Default is 'src'.
+         out.dir
+             The name of the output directory. Default is 'bin'.
+
+         For other overridable properties, look at the beginning of the rules
+         files in the SDK, at tools/ant/build.xml
+
+         Properties related to the SDK location or the project target should
+         be updated using the 'android' tool with the 'update' action.
+
+         This file is an integral part of the build system for your
+         application and should be checked into Version Control Systems.
+
+         -->
+    <property file="ant.properties" />
+
+    <!-- if sdk.dir was not set from one of the property file, then
+         get it from the ANDROID_HOME env var.
+         This must be done before we load project.properties since
+         the proguard config can use sdk.dir -->
+    <property environment="env" />
+    <condition property="sdk.dir" value="${env.ANDROID_SDK_ROOT}">
+        <isset property="env.ANDROID_SDK_ROOT" />
+    </condition>
+
+    <!-- The project.properties file is created and updated by the 'android'
+         tool, as well as ADT.
+
+         This contains project specific properties such as project target, and library
+         dependencies. Lower level build properties are stored in ant.properties
+         (or in .classpath for Eclipse projects).
+
+         This file is an integral part of the build system for your
+         application and should be checked into Version Control Systems. -->
+    <loadproperties srcFile="project.properties" />
+
+    <!-- quick check on sdk.dir -->
+    <fail
+            message="sdk.dir is missing. Make sure to generate local.properties using 'android update project' or to inject it through the ANDROID_HOME environment variable."
+            unless="sdk.dir"
+    />
+
+    <!--
+        Import per project custom build rules if present at the root of the project.
+        This is the place to put custom intermediary targets such as:
+            -pre-build
+            -pre-compile
+            -post-compile (This is typically used for code obfuscation.
+                           Compiled code location: ${out.classes.absolute.dir}
+                           If this is not done in place, override ${out.dex.input.absolute.dir})
+            -post-package
+            -post-build
+            -pre-clean
+    -->
+    <import file="custom_rules.xml" optional="true" />
+
+    <!-- Import the actual build file.
+
+         To customize existing targets, there are two options:
+         - Customize only one target:
+             - copy/paste the target into this file, *before* the
+               <import> task.
+             - customize it to your needs.
+         - Customize the whole content of build.xml
+             - copy/paste the content of the rules files (minus the top node)
+               into this file, replacing the <import> task.
+             - customize to your needs.
+
+         ***********************
+         ****** IMPORTANT ******
+         ***********************
+         In all cases you must update the value of version-tag below to read 'custom' instead of an integer,
+         in order to avoid having your file be overridden by tools such as "android update project"
+    -->
+    <!-- version-tag: 1 -->
+    <import file="${sdk.dir}/tools/ant/build.xml" />
+
+</project>
diff --git a/third_party/libwebrtc/examples/androidtests/project.properties b/third_party/libwebrtc/examples/androidtests/project.properties
new file mode 100644
index 0000000000..a6ca533fe3
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidtests/project.properties
@@ -0,0 +1,16 @@
+# This file is automatically generated by Android Tools.
+# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
+#
+# This file must be checked in Version Control Systems.
+#
+# To customize properties used by the Ant build system edit
+# "ant.properties", and override values to adapt the script to your
+# project structure.
+#
+# To enable ProGuard to shrink and obfuscate your code, uncomment this (available properties: sdk.dir, user.home):
+#proguard.config=${sdk.dir}/tools/proguard/proguard-android.txt:proguard-project.txt
+
+# Project target.
+target=android-22
+
+java.compilerargs=-Xlint:all -Werror
diff --git a/third_party/libwebrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java b/third_party/libwebrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
new file mode 100644
index 0000000000..051d7379bd
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
@@ -0,0 +1,637 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.appspot.apprtc.test;
+
+import static org.junit.Assert.assertTrue;
+import static org.junit.Assert.fail;
+
+import android.os.Build;
+import android.support.test.InstrumentationRegistry;
+import android.support.test.runner.AndroidJUnit4;
+import android.util.Log;
+import androidx.test.filters.SmallTest;
+import java.util.ArrayList;
+import java.util.List;
+import java.util.concurrent.CountDownLatch;
+import java.util.concurrent.ExecutorService;
+import java.util.concurrent.Executors;
+import java.util.concurrent.TimeUnit;
+import org.appspot.apprtc.AppRTCClient.SignalingParameters;
+import org.appspot.apprtc.PeerConnectionClient;
+import org.appspot.apprtc.PeerConnectionClient.PeerConnectionEvents;
+import org.appspot.apprtc.PeerConnectionClient.PeerConnectionParameters;
+import org.junit.After;
+import org.junit.Before;
+import org.junit.Test;
+import org.junit.runner.RunWith;
+import org.webrtc.Camera1Enumerator;
+import org.webrtc.Camera2Enumerator;
+import org.webrtc.CameraEnumerator;
+import org.webrtc.EglBase;
+import org.webrtc.IceCandidate;
+import org.webrtc.PeerConnection;
+import org.webrtc.PeerConnectionFactory;
+import org.webrtc.RTCStatsReport;
+import org.webrtc.SessionDescription;
+import org.webrtc.VideoCapturer;
+import org.webrtc.VideoFrame;
+import org.webrtc.VideoSink;
+
+@RunWith(AndroidJUnit4.class)
+public class PeerConnectionClientTest implements PeerConnectionEvents {
+  private static final String TAG = "RTCClientTest";
+  private static final int ICE_CONNECTION_WAIT_TIMEOUT = 10000;
+  private static final int WAIT_TIMEOUT = 7000;
+  private static final int CAMERA_SWITCH_ATTEMPTS = 3;
+  private static final int VIDEO_RESTART_ATTEMPTS = 3;
+  private static final int CAPTURE_FORMAT_CHANGE_ATTEMPTS = 3;
+  private static final int VIDEO_RESTART_TIMEOUT = 500;
+  private static final int EXPECTED_VIDEO_FRAMES = 10;
+  private static final String VIDEO_CODEC_VP8 = "VP8";
+  private static final String VIDEO_CODEC_VP9 = "VP9";
+  private static final String VIDEO_CODEC_H264 = "H264";
+  private static final int AUDIO_RUN_TIMEOUT = 1000;
+  private static final String LOCAL_RENDERER_NAME = "Local renderer";
+  private static final String REMOTE_RENDERER_NAME = "Remote renderer";
+
+  private static final int MAX_VIDEO_FPS = 30;
+  private static final int WIDTH_VGA = 640;
+  private static final int HEIGHT_VGA = 480;
+  private static final int WIDTH_QVGA = 320;
+  private static final int HEIGHT_QVGA = 240;
+
+  // The peer connection client is assumed to be thread safe in itself; the
+  // reference is written by the test thread and read by worker threads.
+  private volatile PeerConnectionClient pcClient;
+  private volatile boolean loopback;
+
+  // These are protected by their respective event objects.
+  private ExecutorService signalingExecutor;
+  private boolean isClosed;
+  private boolean isIceConnected;
+  private SessionDescription localDesc;
+  private List<IceCandidate> iceCandidates = new ArrayList<>();
+  private final Object localDescEvent = new Object();
+  private final Object iceCandidateEvent = new Object();
+  private final Object iceConnectedEvent = new Object();
+  private final Object closeEvent = new Object();
+
+  // Mock VideoSink implementation.
+  private static class MockSink implements VideoSink {
+    // These are protected by 'this' since we gets called from worker threads.
+    private String rendererName;
+    private boolean renderFrameCalled;
+
+    // Thread-safe in itself.
+    private CountDownLatch doneRendering;
+
+    public MockSink(int expectedFrames, String rendererName) {
+      this.rendererName = rendererName;
+      reset(expectedFrames);
+    }
+
+    // Resets render to wait for new amount of video frames.
+    // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+    @SuppressWarnings("NoSynchronizedMethodCheck")
+    public synchronized void reset(int expectedFrames) {
+      renderFrameCalled = false;
+      doneRendering = new CountDownLatch(expectedFrames);
+    }
+
+    @Override
+    // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+    @SuppressWarnings("NoSynchronizedMethodCheck")
+    public synchronized void onFrame(VideoFrame frame) {
+      if (!renderFrameCalled) {
+        if (rendererName != null) {
+          Log.d(TAG,
+              rendererName + " render frame: " + frame.getRotatedWidth() + " x "
+                  + frame.getRotatedHeight());
+        } else {
+          Log.d(TAG, "Render frame: " + frame.getRotatedWidth() + " x " + frame.getRotatedHeight());
+        }
+      }
+      renderFrameCalled = true;
+      doneRendering.countDown();
+    }
+
+    // This method shouldn't hold any locks or touch member variables since it
+    // blocks.
+    public boolean waitForFramesRendered(int timeoutMs) throws InterruptedException {
+      doneRendering.await(timeoutMs, TimeUnit.MILLISECONDS);
+      return (doneRendering.getCount() <= 0);
+    }
+  }
+
+  // Peer connection events implementation.
+  @Override
+  public void onLocalDescription(SessionDescription desc) {
+    Log.d(TAG, "Local description type: " + desc.type);
+    synchronized (localDescEvent) {
+      localDesc = desc;
+      localDescEvent.notifyAll();
+    }
+  }
+
+  @Override
+  public void onIceCandidate(final IceCandidate candidate) {
+    synchronized (iceCandidateEvent) {
+      Log.d(TAG, "IceCandidate #" + iceCandidates.size() + " : " + candidate.toString());
+      if (loopback) {
+        // Loopback local ICE candidate in a separate thread to avoid adding
+        // remote ICE candidate in a local ICE candidate callback.
+        signalingExecutor.execute(new Runnable() {
+          @Override
+          public void run() {
+            pcClient.addRemoteIceCandidate(candidate);
+          }
+        });
+      }
+      iceCandidates.add(candidate);
+      iceCandidateEvent.notifyAll();
+    }
+  }
+
+  @Override
+  public void onIceCandidatesRemoved(final IceCandidate[] candidates) {
+    // TODO(honghaiz): Add this for tests.
+  }
+
+  @Override
+  public void onIceConnected() {
+    Log.d(TAG, "ICE Connected");
+    synchronized (iceConnectedEvent) {
+      isIceConnected = true;
+      iceConnectedEvent.notifyAll();
+    }
+  }
+
+  @Override
+  public void onIceDisconnected() {
+    Log.d(TAG, "ICE Disconnected");
+    synchronized (iceConnectedEvent) {
+      isIceConnected = false;
+      iceConnectedEvent.notifyAll();
+    }
+  }
+
+  @Override
+  public void onConnected() {
+    Log.d(TAG, "DTLS Connected");
+  }
+
+  @Override
+  public void onDisconnected() {
+    Log.d(TAG, "DTLS Disconnected");
+  }
+
+  @Override
+  public void onPeerConnectionClosed() {
+    Log.d(TAG, "PeerConnection closed");
+    synchronized (closeEvent) {
+      isClosed = true;
+      closeEvent.notifyAll();
+    }
+  }
+
+  @Override
+  public void onPeerConnectionError(String description) {
+    fail("PC Error: " + description);
+  }
+
+  @Override
+  public void onPeerConnectionStatsReady(final RTCStatsReport report) {}
+
+  // Helper wait functions.
+  private boolean waitForLocalDescription(int timeoutMs) throws InterruptedException {
+    synchronized (localDescEvent) {
+      final long endTimeMs = System.currentTimeMillis() + timeoutMs;
+      while (localDesc == null) {
+        final long waitTimeMs = endTimeMs - System.currentTimeMillis();
+        if (waitTimeMs < 0) {
+          return false;
+        }
+        localDescEvent.wait(waitTimeMs);
+      }
+      return true;
+    }
+  }
+
+  private boolean waitForIceCandidates(int timeoutMs) throws InterruptedException {
+    synchronized (iceCandidateEvent) {
+      final long endTimeMs = System.currentTimeMillis() + timeoutMs;
+      while (iceCandidates.size() == 0) {
+        final long waitTimeMs = endTimeMs - System.currentTimeMillis();
+        if (waitTimeMs < 0) {
+          return false;
+        }
+        iceCandidateEvent.wait(timeoutMs);
+      }
+      return true;
+    }
+  }
+
+  private boolean waitForIceConnected(int timeoutMs) throws InterruptedException {
+    synchronized (iceConnectedEvent) {
+      final long endTimeMs = System.currentTimeMillis() + timeoutMs;
+      while (!isIceConnected) {
+        final long waitTimeMs = endTimeMs - System.currentTimeMillis();
+        if (waitTimeMs < 0) {
+          Log.e(TAG, "ICE connection failure");
+          return false;
+        }
+        iceConnectedEvent.wait(timeoutMs);
+      }
+      return true;
+    }
+  }
+
+  private boolean waitForPeerConnectionClosed(int timeoutMs) throws InterruptedException {
+    synchronized (closeEvent) {
+      final long endTimeMs = System.currentTimeMillis() + timeoutMs;
+      while (!isClosed) {
+        final long waitTimeMs = endTimeMs - System.currentTimeMillis();
+        if (waitTimeMs < 0) {
+          return false;
+        }
+        closeEvent.wait(timeoutMs);
+      }
+      return true;
+    }
+  }
+
+  PeerConnectionClient createPeerConnectionClient(MockSink localRenderer, MockSink remoteRenderer,
+      PeerConnectionParameters peerConnectionParameters, VideoCapturer videoCapturer) {
+    List<PeerConnection.IceServer> iceServers = new ArrayList<>();
+    SignalingParameters signalingParameters =
+        new SignalingParameters(iceServers, true, // iceServers, initiator.
+            null, null, null, // clientId, wssUrl, wssPostUrl.
+            null, null); // offerSdp, iceCandidates.
+
+    final EglBase eglBase = EglBase.create();
+    PeerConnectionClient client =
+        new PeerConnectionClient(InstrumentationRegistry.getTargetContext(), eglBase,
+            peerConnectionParameters, this /* PeerConnectionEvents */);
+    PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();
+    options.networkIgnoreMask = 0;
+    options.disableNetworkMonitor = true;
+    client.createPeerConnectionFactory(options);
+    client.createPeerConnection(localRenderer, remoteRenderer, videoCapturer, signalingParameters);
+    client.createOffer();
+    return client;
+  }
+
+  private PeerConnectionParameters createParametersForAudioCall() {
+    return new PeerConnectionParameters(false, /* videoCallEnabled */
+        true, /* loopback */
+        false, /* tracing */
+        // Video codec parameters.
+        0, /* videoWidth */
+        0, /* videoHeight */
+        0, /* videoFps */
+        0, /* videoStartBitrate */
+        "", /* videoCodec */
+        true, /* videoCodecHwAcceleration */
+        false, /* videoFlexfecEnabled */
+        // Audio codec parameters.
+        0, /* audioStartBitrate */
+        "OPUS", /* audioCodec */
+        false, /* noAudioProcessing */
+        false, /* aecDump */
+        false, /* saveInputAudioToFile */
+        false /* useOpenSLES */, false /* disableBuiltInAEC */, false /* disableBuiltInAGC */,
+        false /* disableBuiltInNS */, false /* disableWebRtcAGC */, false /* enableRtcEventLog */,
+        null /* dataChannelParameters */);
+  }
+
+  private VideoCapturer createCameraCapturer(boolean captureToTexture) {
+    final boolean useCamera2 = captureToTexture
+        && Camera2Enumerator.isSupported(InstrumentationRegistry.getTargetContext());
+
+    CameraEnumerator enumerator;
+    if (useCamera2) {
+      enumerator = new Camera2Enumerator(InstrumentationRegistry.getTargetContext());
+    } else {
+      enumerator = new Camera1Enumerator(captureToTexture);
+    }
+    String deviceName = enumerator.getDeviceNames()[0];
+    return enumerator.createCapturer(deviceName, null);
+  }
+
+  private PeerConnectionParameters createParametersForVideoCall(String videoCodec) {
+    return new PeerConnectionParameters(true, /* videoCallEnabled */
+        true, /* loopback */
+        false, /* tracing */
+        // Video codec parameters.
+        0, /* videoWidth */
+        0, /* videoHeight */
+        0, /* videoFps */
+        0, /* videoStartBitrate */
+        videoCodec, /* videoCodec */
+        true, /* videoCodecHwAcceleration */
+        false, /* videoFlexfecEnabled */
+        // Audio codec parameters.
+        0, /* audioStartBitrate */
+        "OPUS", /* audioCodec */
+        false, /* noAudioProcessing */
+        false, /* aecDump */
+        false, /* saveInputAudioToFile */
+        false /* useOpenSLES */, false /* disableBuiltInAEC */, false /* disableBuiltInAGC */,
+        false /* disableBuiltInNS */, false /* disableWebRtcAGC */, false /* enableRtcEventLog */,
+        null /* dataChannelParameters */);
+  }
+
+  @Before
+  public void setUp() {
+    signalingExecutor = Executors.newSingleThreadExecutor();
+  }
+
+  @After
+  public void tearDown() {
+    signalingExecutor.shutdown();
+  }
+
+  @Test
+  @SmallTest
+  public void testSetLocalOfferMakesVideoFlowLocally() throws InterruptedException {
+    Log.d(TAG, "testSetLocalOfferMakesVideoFlowLocally");
+    MockSink localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME);
+    pcClient = createPeerConnectionClient(localRenderer,
+        new MockSink(/* expectedFrames= */ 0, /* rendererName= */ null),
+        createParametersForVideoCall(VIDEO_CODEC_VP8),
+        createCameraCapturer(false /* captureToTexture */));
+
+    // Wait for local description and ice candidates set events.
+    assertTrue("Local description was not set.", waitForLocalDescription(WAIT_TIMEOUT));
+    assertTrue("ICE candidates were not generated.", waitForIceCandidates(WAIT_TIMEOUT));
+
+    // Check that local video frames were rendered.
+    assertTrue(
+        "Local video frames were not rendered.", localRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+
+    pcClient.close();
+    assertTrue(
+        "PeerConnection close event was not received.", waitForPeerConnectionClosed(WAIT_TIMEOUT));
+    Log.d(TAG, "testSetLocalOfferMakesVideoFlowLocally Done.");
+  }
+
+  private void doLoopbackTest(PeerConnectionParameters parameters, VideoCapturer videoCapturer,
+      boolean decodeToTexture) throws InterruptedException {
+    loopback = true;
+    MockSink localRenderer = null;
+    MockSink remoteRenderer = null;
+    if (parameters.videoCallEnabled) {
+      Log.d(TAG, "testLoopback for video " + parameters.videoCodec);
+      localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME);
+      remoteRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME);
+    } else {
+      Log.d(TAG, "testLoopback for audio.");
+    }
+    pcClient = createPeerConnectionClient(localRenderer, remoteRenderer, parameters, videoCapturer);
+
+    // Wait for local description, change type to answer and set as remote description.
+    assertTrue("Local description was not set.", waitForLocalDescription(WAIT_TIMEOUT));
+    SessionDescription remoteDescription = new SessionDescription(
+        SessionDescription.Type.fromCanonicalForm("answer"), localDesc.description);
+    pcClient.setRemoteDescription(remoteDescription);
+
+    // Wait for ICE connection.
+    assertTrue("ICE connection failure.", waitForIceConnected(ICE_CONNECTION_WAIT_TIMEOUT));
+
+    if (parameters.videoCallEnabled) {
+      // Check that local and remote video frames were rendered.
+      assertTrue("Local video frames were not rendered.",
+          localRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+      assertTrue("Remote video frames were not rendered.",
+          remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+    } else {
+      // For audio just sleep for 1 sec.
+      // TODO(glaznev): check how we can detect that remote audio was rendered.
+      Thread.sleep(AUDIO_RUN_TIMEOUT);
+    }
+
+    pcClient.close();
+    assertTrue(waitForPeerConnectionClosed(WAIT_TIMEOUT));
+    Log.d(TAG, "testLoopback done.");
+  }
+
+  @Test
+  @SmallTest
+  public void testLoopbackAudio() throws InterruptedException {
+    doLoopbackTest(createParametersForAudioCall(), null, false /* decodeToTexture */);
+  }
+
+  @Test
+  @SmallTest
+  public void testLoopbackVp8() throws InterruptedException {
+    doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_VP8),
+        createCameraCapturer(false /* captureToTexture */), false /* decodeToTexture */);
+  }
+
+  @Test
+  @SmallTest
+  public void testLoopbackVp9() throws InterruptedException {
+    doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_VP9),
+        createCameraCapturer(false /* captureToTexture */), false /* decodeToTexture */);
+  }
+
+  @Test
+  @SmallTest
+  public void testLoopbackH264() throws InterruptedException {
+    doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_H264),
+        createCameraCapturer(false /* captureToTexture */), false /* decodeToTexture */);
+  }
+
+  @Test
+  @SmallTest
+  public void testLoopbackVp8DecodeToTexture() throws InterruptedException {
+    doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_VP8),
+        createCameraCapturer(false /* captureToTexture */), true /* decodeToTexture */);
+  }
+
+  @Test
+  @SmallTest
+  public void testLoopbackVp9DecodeToTexture() throws InterruptedException {
+    doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_VP9),
+        createCameraCapturer(false /* captureToTexture */), true /* decodeToTexture */);
+  }
+
+  @Test
+  @SmallTest
+  public void testLoopbackH264DecodeToTexture() throws InterruptedException {
+    doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_H264),
+        createCameraCapturer(false /* captureToTexture */), true /* decodeToTexture */);
+  }
+
+  @Test
+  @SmallTest
+  public void testLoopbackVp8CaptureToTexture() throws InterruptedException {
+    doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_VP8),
+        createCameraCapturer(true /* captureToTexture */), true /* decodeToTexture */);
+  }
+
+  @Test
+  @SmallTest
+  public void testLoopbackH264CaptureToTexture() throws InterruptedException {
+    doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_H264),
+        createCameraCapturer(true /* captureToTexture */), true /* decodeToTexture */);
+  }
+
+  // Checks if default front camera can be switched to back camera and then
+  // again to front camera.
+  @Test
+  @SmallTest
+  public void testCameraSwitch() throws InterruptedException {
+    Log.d(TAG, "testCameraSwitch");
+    loopback = true;
+
+    MockSink localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME);
+    MockSink remoteRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME);
+
+    pcClient = createPeerConnectionClient(localRenderer, remoteRenderer,
+        createParametersForVideoCall(VIDEO_CODEC_VP8),
+        createCameraCapturer(false /* captureToTexture */));
+
+    // Wait for local description, set type to answer and set as remote description.
+    assertTrue("Local description was not set.", waitForLocalDescription(WAIT_TIMEOUT));
+    SessionDescription remoteDescription = new SessionDescription(
+        SessionDescription.Type.fromCanonicalForm("answer"), localDesc.description);
+    pcClient.setRemoteDescription(remoteDescription);
+
+    // Wait for ICE connection.
+    assertTrue("ICE connection failure.", waitForIceConnected(ICE_CONNECTION_WAIT_TIMEOUT));
+
+    // Check that local and remote video frames were rendered.
+    assertTrue("Local video frames were not rendered before camera switch.",
+        localRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+    assertTrue("Remote video frames were not rendered before camera switch.",
+        remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+
+    for (int i = 0; i < CAMERA_SWITCH_ATTEMPTS; i++) {
+      // Try to switch camera
+      pcClient.switchCamera();
+
+      // Reset video renders and check that local and remote video frames
+      // were rendered after camera switch.
+      localRenderer.reset(EXPECTED_VIDEO_FRAMES);
+      remoteRenderer.reset(EXPECTED_VIDEO_FRAMES);
+      assertTrue("Local video frames were not rendered after camera switch.",
+          localRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+      assertTrue("Remote video frames were not rendered after camera switch.",
+          remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+    }
+    pcClient.close();
+    assertTrue(waitForPeerConnectionClosed(WAIT_TIMEOUT));
+    Log.d(TAG, "testCameraSwitch done.");
+  }
+
+  // Checks if video source can be restarted - simulate app goes to
+  // background and back to foreground.
+  @Test
+  @SmallTest
+  public void testVideoSourceRestart() throws InterruptedException {
+    Log.d(TAG, "testVideoSourceRestart");
+    loopback = true;
+
+    MockSink localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME);
+    MockSink remoteRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME);
+
+    pcClient = createPeerConnectionClient(localRenderer, remoteRenderer,
+        createParametersForVideoCall(VIDEO_CODEC_VP8),
+        createCameraCapturer(false /* captureToTexture */));
+
+    // Wait for local description, set type to answer and set as remote description.
+    assertTrue("Local description was not set.", waitForLocalDescription(WAIT_TIMEOUT));
+    SessionDescription remoteDescription = new SessionDescription(
+        SessionDescription.Type.fromCanonicalForm("answer"), localDesc.description);
+    pcClient.setRemoteDescription(remoteDescription);
+
+    // Wait for ICE connection.
+    assertTrue("ICE connection failure.", waitForIceConnected(ICE_CONNECTION_WAIT_TIMEOUT));
+
+    // Check that local and remote video frames were rendered.
+    assertTrue("Local video frames were not rendered before video restart.",
+        localRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+    assertTrue("Remote video frames were not rendered before video restart.",
+        remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+
+    // Stop and then start video source a few times.
+    for (int i = 0; i < VIDEO_RESTART_ATTEMPTS; i++) {
+      pcClient.stopVideoSource();
+      Thread.sleep(VIDEO_RESTART_TIMEOUT);
+      pcClient.startVideoSource();
+
+      // Reset video renders and check that local and remote video frames
+      // were rendered after video restart.
+      localRenderer.reset(EXPECTED_VIDEO_FRAMES);
+      remoteRenderer.reset(EXPECTED_VIDEO_FRAMES);
+      assertTrue("Local video frames were not rendered after video restart.",
+          localRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+      assertTrue("Remote video frames were not rendered after video restart.",
+          remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+    }
+    pcClient.close();
+    assertTrue(waitForPeerConnectionClosed(WAIT_TIMEOUT));
+    Log.d(TAG, "testVideoSourceRestart done.");
+  }
+
+  // Checks if capture format can be changed on fly and decoder can be reset properly.
+  @Test
+  @SmallTest
+  public void testCaptureFormatChange() throws InterruptedException {
+    Log.d(TAG, "testCaptureFormatChange");
+    loopback = true;
+
+    MockSink localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME);
+    MockSink remoteRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME);
+
+    pcClient = createPeerConnectionClient(localRenderer, remoteRenderer,
+        createParametersForVideoCall(VIDEO_CODEC_VP8),
+        createCameraCapturer(false /* captureToTexture */));
+
+    // Wait for local description, set type to answer and set as remote description.
+    assertTrue("Local description was not set.", waitForLocalDescription(WAIT_TIMEOUT));
+    SessionDescription remoteDescription = new SessionDescription(
+        SessionDescription.Type.fromCanonicalForm("answer"), localDesc.description);
+    pcClient.setRemoteDescription(remoteDescription);
+
+    // Wait for ICE connection.
+    assertTrue("ICE connection failure.", waitForIceConnected(ICE_CONNECTION_WAIT_TIMEOUT));
+
+    // Check that local and remote video frames were rendered.
+    assertTrue("Local video frames were not rendered before camera resolution change.",
+        localRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+    assertTrue("Remote video frames were not rendered before camera resolution change.",
+        remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+
+    // Change capture output format a few times.
+    for (int i = 0; i < 2 * CAPTURE_FORMAT_CHANGE_ATTEMPTS; i++) {
+      if (i % 2 == 0) {
+        pcClient.changeCaptureFormat(WIDTH_VGA, HEIGHT_VGA, MAX_VIDEO_FPS);
+      } else {
+        pcClient.changeCaptureFormat(WIDTH_QVGA, HEIGHT_QVGA, MAX_VIDEO_FPS);
+      }
+
+      // Reset video renders and check that local and remote video frames
+      // were rendered after capture format change.
+      localRenderer.reset(EXPECTED_VIDEO_FRAMES);
+      remoteRenderer.reset(EXPECTED_VIDEO_FRAMES);
+      assertTrue("Local video frames were not rendered after capture format change.",
+          localRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+      assertTrue("Remote video frames were not rendered after capture format change.",
+          remoteRenderer.waitForFramesRendered(WAIT_TIMEOUT));
+    }
+
+    pcClient.close();
+    assertTrue(waitForPeerConnectionClosed(WAIT_TIMEOUT));
+    Log.d(TAG, "testCaptureFormatChange done.");
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidtests/third_party/.gitignore b/third_party/libwebrtc/examples/androidtests/third_party/.gitignore
new file mode 100644
index 0000000000..52acefb2ec
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidtests/third_party/.gitignore
@@ -0,0 +1,3 @@
+# This file is needed for projects that has this directory as a separate Git
+# mirror in DEPS. Without it, a lot is wiped and re-downloaded for each sync.
+/gradle
diff --git a/third_party/libwebrtc/examples/androidtests/third_party/README.webrtc b/third_party/libwebrtc/examples/androidtests/third_party/README.webrtc
new file mode 100644
index 0000000000..a6ea884923
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidtests/third_party/README.webrtc
@@ -0,0 +1,10 @@
+The third_party directory contains sources from other projects.
+
+Code in third_party must document the license under which the source is being
+used. If the source itself does not include a license header or file, create
+an entry in this file that refers to reliable documentation of the project's
+license terms on the web (and add a note pointing here in the README file in
+that directory).
+
+<Include table of license information here, once it is available>
+
diff --git a/third_party/libwebrtc/examples/androidvoip/AndroidManifest.xml b/third_party/libwebrtc/examples/androidvoip/AndroidManifest.xml
new file mode 100644
index 0000000000..8e096b0452
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidvoip/AndroidManifest.xml
@@ -0,0 +1,38 @@
+<?xml version="1.0" encoding="utf-8"?>
+
+<!--
+ *  Copyright 2020 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+-->
+
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+  package="org.webrtc.examples.androidvoip">
+
+
+  <uses-permission android:name="android.permission.INTERNET" />
+  <uses-permission android:name="android.permission.RECORD_AUDIO" />
+  <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
+
+  <uses-feature android:name="android.hardware.microphone" android:required="true" />
+  <uses-feature android:name="android.hardware.telephony" android:required="false" />
+
+  <application
+    android:allowBackup="true"
+    android:label="@string/app_name"
+    android:supportsRtl="true">
+    <activity android:name=".MainActivity"
+      android:windowSoftInputMode="stateHidden"
+      android:exported="true">
+      <intent-filter>
+        <action android:name="android.intent.action.MAIN" />
+        <category android:name="android.intent.category.LAUNCHER" />
+      </intent-filter>
+    </activity>
+  </application>
+
+</manifest>
diff --git a/third_party/libwebrtc/examples/androidvoip/BUILD.gn b/third_party/libwebrtc/examples/androidvoip/BUILD.gn
new file mode 100644
index 0000000000..3d5186f279
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidvoip/BUILD.gn
@@ -0,0 +1,95 @@
+# Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("//webrtc.gni")
+
+if (is_android) {
+  rtc_android_apk("androidvoip") {
+    testonly = true
+    apk_name = "androidvoip"
+    android_manifest = "AndroidManifest.xml"
+    min_sdk_version = 21
+    target_sdk_version = 31
+
+    sources = [
+      "java/org/webrtc/examples/androidvoip/MainActivity.java",
+      "java/org/webrtc/examples/androidvoip/OnVoipClientTaskCompleted.java",
+      "java/org/webrtc/examples/androidvoip/VoipClient.java",
+    ]
+
+    deps = [
+      ":resources",
+      "//modules/audio_device:audio_device_java",
+      "//rtc_base:base_java",
+      "//sdk/android:base_java",
+      "//sdk/android:java_audio_device_module_java",
+      "//sdk/android:video_java",
+      "//third_party/androidx:androidx_core_core_java",
+      "//third_party/androidx:androidx_legacy_legacy_support_v4_java",
+    ]
+
+    shared_libraries = [ ":examples_androidvoip_jni" ]
+  }
+
+  generate_jni("generated_jni") {
+    testonly = true
+    sources = [ "java/org/webrtc/examples/androidvoip/VoipClient.java" ]
+    namespace = "webrtc_examples"
+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
+  }
+
+  rtc_shared_library("examples_androidvoip_jni") {
+    testonly = true
+    sources = [
+      "jni/android_voip_client.cc",
+      "jni/android_voip_client.h",
+      "jni/onload.cc",
+    ]
+
+    suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
+    configs += [ "//build/config/android:hide_all_but_jni" ]
+
+    deps = [
+      ":generated_jni",
+      "../../rtc_base:async_packet_socket",
+      "../../rtc_base:async_udp_socket",
+      "../../rtc_base:logging",
+      "../../rtc_base:network",
+      "../../rtc_base:socket_address",
+      "../../rtc_base:socket_server",
+      "../../rtc_base:ssl",
+      "../../rtc_base:threading",
+      "//api:transport_api",
+      "//api/audio_codecs:audio_codecs_api",
+      "//api/audio_codecs:builtin_audio_decoder_factory",
+      "//api/audio_codecs:builtin_audio_encoder_factory",
+      "//api/task_queue:default_task_queue_factory",
+      "//api/voip:voip_api",
+      "//api/voip:voip_engine_factory",
+      "//rtc_base/third_party/sigslot:sigslot",
+      "//sdk/android:native_api_audio_device_module",
+      "//sdk/android:native_api_base",
+      "//sdk/android:native_api_jni",
+      "//third_party/abseil-cpp/absl/memory:memory",
+    ]
+  }
+
+  android_resources("resources") {
+    testonly = true
+    custom_package = "org.webrtc.examples.androidvoip"
+    sources = [
+      "res/layout/activity_main.xml",
+      "res/values/colors.xml",
+      "res/values/strings.xml",
+    ]
+
+    # Needed for Bazel converter.
+    resource_dirs = [ "res" ]
+    assert(resource_dirs != [])  # Mark as used.
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidvoip/DEPS b/third_party/libwebrtc/examples/androidvoip/DEPS
new file mode 100644
index 0000000000..edb714dd44
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidvoip/DEPS
@@ -0,0 +1,3 @@
+include_rules = [
+  "+sdk/android/native_api",
+]
diff --git a/third_party/libwebrtc/examples/androidvoip/OWNERS b/third_party/libwebrtc/examples/androidvoip/OWNERS
new file mode 100644
index 0000000000..e7d3200562
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidvoip/OWNERS
@@ -0,0 +1,2 @@
+natim@webrtc.org
+xalep@webrtc.org
diff --git a/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/MainActivity.java b/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/MainActivity.java
new file mode 100644
index 0000000000..d06d6adf0d
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/MainActivity.java
@@ -0,0 +1,341 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.examples.androidvoip;
+
+import android.Manifest.permission;
+import android.app.Activity;
+import android.app.AlertDialog;
+import android.content.Context;
+import android.content.pm.PackageManager;
+import android.os.Bundle;
+import android.view.Gravity;
+import android.view.View;
+import android.widget.AdapterView;
+import android.widget.ArrayAdapter;
+import android.widget.Button;
+import android.widget.EditText;
+import android.widget.RelativeLayout;
+import android.widget.ScrollView;
+import android.widget.Spinner;
+import android.widget.Switch;
+import android.widget.TextView;
+import android.widget.Toast;
+import android.widget.ToggleButton;
+import androidx.core.app.ActivityCompat;
+import androidx.core.content.ContextCompat;
+import java.util.ArrayList;
+import java.util.Arrays;
+import java.util.HashSet;
+import java.util.List;
+import java.util.Set;
+import java.util.stream.Collectors;
+import org.webrtc.ContextUtils;
+
+public class MainActivity extends Activity implements OnVoipClientTaskCompleted {
+  private static final int NUM_SUPPORTED_CODECS = 6;
+
+  private VoipClient voipClient;
+  private List<String> supportedCodecs;
+  private boolean[] isDecoderSelected;
+  private Set<Integer> selectedDecoders;
+
+  private Toast toast;
+  private ScrollView scrollView;
+  private TextView localIPAddressTextView;
+  private EditText localPortNumberEditText;
+  private EditText remoteIPAddressEditText;
+  private EditText remotePortNumberEditText;
+  private Spinner encoderSpinner;
+  private Button decoderSelectionButton;
+  private TextView decodersTextView;
+  private ToggleButton sessionButton;
+  private RelativeLayout switchLayout;
+  private Switch sendSwitch;
+  private Switch playoutSwitch;
+
+  @Override
+  protected void onCreate(Bundle savedInstance) {
+    ContextUtils.initialize(getApplicationContext());
+
+    super.onCreate(savedInstance);
+    setContentView(R.layout.activity_main);
+
+    System.loadLibrary("examples_androidvoip_jni");
+
+    voipClient = new VoipClient(getApplicationContext(), this);
+    voipClient.getAndSetUpLocalIPAddress();
+    voipClient.getAndSetUpSupportedCodecs();
+
+    isDecoderSelected = new boolean[NUM_SUPPORTED_CODECS];
+    selectedDecoders = new HashSet<>();
+
+    toast = Toast.makeText(this, "", Toast.LENGTH_SHORT);
+
+    scrollView = (ScrollView) findViewById(R.id.scroll_view);
+    localIPAddressTextView = (TextView) findViewById(R.id.local_ip_address_text_view);
+    localPortNumberEditText = (EditText) findViewById(R.id.local_port_number_edit_text);
+    remoteIPAddressEditText = (EditText) findViewById(R.id.remote_ip_address_edit_text);
+    remotePortNumberEditText = (EditText) findViewById(R.id.remote_port_number_edit_text);
+    encoderSpinner = (Spinner) findViewById(R.id.encoder_spinner);
+    decoderSelectionButton = (Button) findViewById(R.id.decoder_selection_button);
+    decodersTextView = (TextView) findViewById(R.id.decoders_text_view);
+    sessionButton = (ToggleButton) findViewById(R.id.session_button);
+    switchLayout = (RelativeLayout) findViewById(R.id.switch_layout);
+    sendSwitch = (Switch) findViewById(R.id.start_send_switch);
+    playoutSwitch = (Switch) findViewById(R.id.start_playout_switch);
+
+    setUpSessionButton();
+    setUpSendAndPlayoutSwitch();
+  }
+
+  private void setUpEncoderSpinner(List<String> supportedCodecs) {
+    ArrayAdapter<String> encoderAdapter =
+        new ArrayAdapter<String>(this, android.R.layout.simple_spinner_item, supportedCodecs);
+    encoderAdapter.setDropDownViewResource(android.R.layout.simple_spinner_dropdown_item);
+    encoderSpinner.setAdapter(encoderAdapter);
+    encoderSpinner.setOnItemSelectedListener(new AdapterView.OnItemSelectedListener() {
+      @Override
+      public void onItemSelected(AdapterView<?> parent, View view, int position, long id) {
+        voipClient.setEncoder((String) parent.getSelectedItem());
+      }
+      @Override
+      public void onNothingSelected(AdapterView<?> parent) {}
+    });
+  }
+
+  private List<String> getSelectedDecoders() {
+    List<String> decoders = new ArrayList<>();
+    for (int i = 0; i < supportedCodecs.size(); i++) {
+      if (selectedDecoders.contains(i)) {
+        decoders.add(supportedCodecs.get(i));
+      }
+    }
+    return decoders;
+  }
+
+  private void setUpDecoderSelectionButton(List<String> supportedCodecs) {
+    decoderSelectionButton.setOnClickListener((view) -> {
+      AlertDialog.Builder dialogBuilder = new AlertDialog.Builder(this);
+      dialogBuilder.setTitle(R.string.dialog_title);
+
+      // Populate multi choice items with supported decoders.
+      String[] supportedCodecsArray = supportedCodecs.toArray(new String[0]);
+      dialogBuilder.setMultiChoiceItems(
+          supportedCodecsArray, isDecoderSelected, (dialog, position, isChecked) -> {
+            if (isChecked) {
+              selectedDecoders.add(position);
+            } else if (!isChecked) {
+              selectedDecoders.remove(position);
+            }
+          });
+
+      // "Ok" button.
+      dialogBuilder.setPositiveButton(R.string.ok_label, (dialog, position) -> {
+        List<String> decoders = getSelectedDecoders();
+        String result = decoders.stream().collect(Collectors.joining(", "));
+        if (result.isEmpty()) {
+          decodersTextView.setText(R.string.decoders_text_view_default);
+        } else {
+          decodersTextView.setText(result);
+        }
+        voipClient.setDecoders(decoders);
+      });
+
+      // "Dismiss" button.
+      dialogBuilder.setNegativeButton(
+          R.string.dismiss_label, (dialog, position) -> { dialog.dismiss(); });
+
+      // "Clear All" button.
+      dialogBuilder.setNeutralButton(R.string.clear_all_label, (dialog, position) -> {
+        Arrays.fill(isDecoderSelected, false);
+        selectedDecoders.clear();
+        decodersTextView.setText(R.string.decoders_text_view_default);
+      });
+
+      AlertDialog dialog = dialogBuilder.create();
+      dialog.show();
+    });
+  }
+
+  private void setUpSessionButton() {
+    sessionButton.setOnCheckedChangeListener((button, isChecked) -> {
+      // Ask for permission on RECORD_AUDIO if not granted.
+      if (ContextCompat.checkSelfPermission(this, permission.RECORD_AUDIO)
+          != PackageManager.PERMISSION_GRANTED) {
+        String[] sList = {permission.RECORD_AUDIO};
+        ActivityCompat.requestPermissions(this, sList, 1);
+      }
+
+      if (isChecked) {
+        // Order matters here, addresses have to be set before starting session
+        // before setting codec.
+        voipClient.setLocalAddress(localIPAddressTextView.getText().toString(),
+            Integer.parseInt(localPortNumberEditText.getText().toString()));
+        voipClient.setRemoteAddress(remoteIPAddressEditText.getText().toString(),
+            Integer.parseInt(remotePortNumberEditText.getText().toString()));
+        voipClient.startSession();
+        voipClient.setEncoder((String) encoderSpinner.getSelectedItem());
+        voipClient.setDecoders(getSelectedDecoders());
+      } else {
+        voipClient.stopSession();
+      }
+    });
+  }
+
+  private void setUpSendAndPlayoutSwitch() {
+    sendSwitch.setOnCheckedChangeListener((button, isChecked) -> {
+      if (isChecked) {
+        voipClient.startSend();
+      } else {
+        voipClient.stopSend();
+      }
+    });
+
+    playoutSwitch.setOnCheckedChangeListener((button, isChecked) -> {
+      if (isChecked) {
+        voipClient.startPlayout();
+      } else {
+        voipClient.stopPlayout();
+      }
+    });
+  }
+
+  private void setUpIPAddressEditTexts(String localIPAddress) {
+    if (localIPAddress.isEmpty()) {
+      showToast("Please check your network configuration");
+    } else {
+      localIPAddressTextView.setText(localIPAddress);
+      // By default remote IP address is the same as local IP address.
+      remoteIPAddressEditText.setText(localIPAddress);
+    }
+  }
+
+  private void showToast(String message) {
+    if (toast != null) {
+      toast.cancel();
+      toast = Toast.makeText(this, message, Toast.LENGTH_SHORT);
+      toast.setGravity(Gravity.TOP, 0, 200);
+      toast.show();
+    }
+  }
+
+  @Override
+  protected void onDestroy() {
+    voipClient.close();
+    voipClient = null;
+
+    super.onDestroy();
+  }
+
+  @Override
+  public void onGetLocalIPAddressCompleted(String localIPAddress) {
+    runOnUiThread(() -> { setUpIPAddressEditTexts(localIPAddress); });
+  }
+
+  @Override
+  public void onGetSupportedCodecsCompleted(List<String> supportedCodecs) {
+    runOnUiThread(() -> {
+      this.supportedCodecs = supportedCodecs;
+      setUpEncoderSpinner(supportedCodecs);
+      setUpDecoderSelectionButton(supportedCodecs);
+    });
+  }
+
+  @Override
+  public void onVoipClientInitializationCompleted(boolean isSuccessful) {
+    runOnUiThread(() -> {
+      if (!isSuccessful) {
+        showToast("Error initializing audio device");
+      }
+    });
+  }
+
+  @Override
+  public void onStartSessionCompleted(boolean isSuccessful) {
+    runOnUiThread(() -> {
+      if (isSuccessful) {
+        showToast("Session started");
+        switchLayout.setVisibility(View.VISIBLE);
+        scrollView.post(() -> { scrollView.fullScroll(ScrollView.FOCUS_DOWN); });
+      } else {
+        showToast("Failed to start session");
+      }
+    });
+  }
+
+  @Override
+  public void onStopSessionCompleted(boolean isSuccessful) {
+    runOnUiThread(() -> {
+      if (isSuccessful) {
+        showToast("Session stopped");
+        // Set listeners to null so the checked state can be changed programmatically.
+        sendSwitch.setOnCheckedChangeListener(null);
+        playoutSwitch.setOnCheckedChangeListener(null);
+        sendSwitch.setChecked(false);
+        playoutSwitch.setChecked(false);
+        // Redo the switch listener setup.
+        setUpSendAndPlayoutSwitch();
+        switchLayout.setVisibility(View.GONE);
+      } else {
+        showToast("Failed to stop session");
+      }
+    });
+  }
+
+  @Override
+  public void onStartSendCompleted(boolean isSuccessful) {
+    runOnUiThread(() -> {
+      if (isSuccessful) {
+        showToast("Started sending");
+      } else {
+        showToast("Error initializing microphone");
+      }
+    });
+  }
+
+  @Override
+  public void onStopSendCompleted(boolean isSuccessful) {
+    runOnUiThread(() -> {
+      if (isSuccessful) {
+        showToast("Stopped sending");
+      } else {
+        showToast("Microphone termination failed");
+      }
+    });
+  }
+
+  @Override
+  public void onStartPlayoutCompleted(boolean isSuccessful) {
+    runOnUiThread(() -> {
+      if (isSuccessful) {
+        showToast("Started playout");
+      } else {
+        showToast("Error initializing speaker");
+      }
+    });
+  }
+
+  @Override
+  public void onStopPlayoutCompleted(boolean isSuccessful) {
+    runOnUiThread(() -> {
+      if (isSuccessful) {
+        showToast("Stopped playout");
+      } else {
+        showToast("Speaker termination failed");
+      }
+    });
+  }
+
+  @Override
+  public void onUninitializedVoipClient() {
+    runOnUiThread(() -> { showToast("Voip client is uninitialized"); });
+  }
+}
diff --git a/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/OnVoipClientTaskCompleted.java b/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/OnVoipClientTaskCompleted.java
new file mode 100644
index 0000000000..bb85e048bb
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/OnVoipClientTaskCompleted.java
@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.examples.androidvoip;
+
+import java.util.List;
+
+public interface OnVoipClientTaskCompleted {
+  void onGetLocalIPAddressCompleted(String localIPAddress);
+  void onGetSupportedCodecsCompleted(List<String> supportedCodecs);
+  void onVoipClientInitializationCompleted(boolean isSuccessful);
+  void onStartSessionCompleted(boolean isSuccessful);
+  void onStopSessionCompleted(boolean isSuccessful);
+  void onStartSendCompleted(boolean isSuccessful);
+  void onStopSendCompleted(boolean isSuccessful);
+  void onStartPlayoutCompleted(boolean isSuccessful);
+  void onStopPlayoutCompleted(boolean isSuccessful);
+  void onUninitializedVoipClient();
+}
diff --git a/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/VoipClient.java b/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/VoipClient.java
new file mode 100644
index 0000000000..69a993d344
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidvoip/java/org/webrtc/examples/androidvoip/VoipClient.java
@@ -0,0 +1,191 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.examples.androidvoip;
+
+import android.content.Context;
+import android.os.Handler;
+import android.os.HandlerThread;
+import java.util.ArrayList;
+import java.util.List;
+import org.webrtc.CalledByNative;
+
+public class VoipClient {
+  private long nativeClient;
+  private OnVoipClientTaskCompleted listener;
+
+  public VoipClient(Context applicationContext, OnVoipClientTaskCompleted listener) {
+    this.listener = listener;
+    nativeClient = nativeCreateClient(applicationContext, this);
+  }
+
+  private boolean isInitialized() {
+    return nativeClient != 0;
+  }
+
+  public void getAndSetUpSupportedCodecs() {
+    if (isInitialized()) {
+      nativeGetSupportedCodecs(nativeClient);
+    } else {
+      listener.onUninitializedVoipClient();
+    }
+  }
+
+  public void getAndSetUpLocalIPAddress() {
+    if (isInitialized()) {
+      nativeGetLocalIPAddress(nativeClient);
+    } else {
+      listener.onUninitializedVoipClient();
+    }
+  }
+
+  public void setEncoder(String encoder) {
+    if (isInitialized()) {
+      nativeSetEncoder(nativeClient, encoder);
+    } else {
+      listener.onUninitializedVoipClient();
+    }
+  }
+
+  public void setDecoders(List<String> decoders) {
+    if (isInitialized()) {
+      nativeSetDecoders(nativeClient, decoders);
+    } else {
+      listener.onUninitializedVoipClient();
+    }
+  }
+
+  public void setLocalAddress(String ipAddress, int portNumber) {
+    if (isInitialized()) {
+      nativeSetLocalAddress(nativeClient, ipAddress, portNumber);
+    } else {
+      listener.onUninitializedVoipClient();
+    }
+  }
+
+  public void setRemoteAddress(String ipAddress, int portNumber) {
+    if (isInitialized()) {
+      nativeSetRemoteAddress(nativeClient, ipAddress, portNumber);
+    } else {
+      listener.onUninitializedVoipClient();
+    }
+  }
+
+  public void startSession() {
+    if (isInitialized()) {
+      nativeStartSession(nativeClient);
+    } else {
+      listener.onUninitializedVoipClient();
+    }
+  }
+
+  public void stopSession() {
+    if (isInitialized()) {
+      nativeStopSession(nativeClient);
+    } else {
+      listener.onUninitializedVoipClient();
+    }
+  }
+
+  public void startSend() {
+    if (isInitialized()) {
+      nativeStartSend(nativeClient);
+    } else {
+      listener.onUninitializedVoipClient();
+    }
+  }
+
+  public void stopSend() {
+    if (isInitialized()) {
+      nativeStopSend(nativeClient);
+    } else {
+      listener.onUninitializedVoipClient();
+    }
+  }
+
+  public void startPlayout() {
+    if (isInitialized()) {
+      nativeStartPlayout(nativeClient);
+    } else {
+      listener.onUninitializedVoipClient();
+    }
+  }
+
+  public void stopPlayout() {
+    if (isInitialized()) {
+      nativeStopPlayout(nativeClient);
+    } else {
+      listener.onUninitializedVoipClient();
+    }
+  }
+
+  public void close() {
+    nativeDelete(nativeClient);
+    nativeClient = 0;
+  }
+
+  @CalledByNative
+  public void onGetLocalIPAddressCompleted(String localIPAddress) {
+    listener.onGetLocalIPAddressCompleted(localIPAddress);
+  }
+
+  @CalledByNative
+  public void onGetSupportedCodecsCompleted(List<String> supportedCodecs) {
+    listener.onGetSupportedCodecsCompleted(supportedCodecs);
+  }
+
+  @CalledByNative
+  public void onStartSessionCompleted(boolean isSuccessful) {
+    listener.onStartSessionCompleted(isSuccessful);
+  }
+
+  @CalledByNative
+  public void onStopSessionCompleted(boolean isSuccessful) {
+    listener.onStopSessionCompleted(isSuccessful);
+  }
+
+  @CalledByNative
+  public void onStartSendCompleted(boolean isSuccessful) {
+    listener.onStartSendCompleted(isSuccessful);
+  }
+
+  @CalledByNative
+  public void onStopSendCompleted(boolean isSuccessful) {
+    listener.onStopSendCompleted(isSuccessful);
+  }
+
+  @CalledByNative
+  public void onStartPlayoutCompleted(boolean isSuccessful) {
+    listener.onStartPlayoutCompleted(isSuccessful);
+  }
+
+  @CalledByNative
+  public void onStopPlayoutCompleted(boolean isSuccessful) {
+    listener.onStopPlayoutCompleted(isSuccessful);
+  }
+
+  private static native long nativeCreateClient(
+      Context applicationContext, VoipClient javaVoipClient);
+  private static native void nativeGetSupportedCodecs(long nativeAndroidVoipClient);
+  private static native void nativeGetLocalIPAddress(long nativeAndroidVoipClient);
+  private static native void nativeSetEncoder(long nativeAndroidVoipClient, String encoder);
+  private static native void nativeSetDecoders(long nativeAndroidVoipClient, List<String> decoders);
+  private static native void nativeSetLocalAddress(
+      long nativeAndroidVoipClient, String ipAddress, int portNumber);
+  private static native void nativeSetRemoteAddress(
+      long nativeAndroidVoipClient, String ipAddress, int portNumber);
+  private static native void nativeStartSession(long nativeAndroidVoipClient);
+  private static native void nativeStopSession(long nativeAndroidVoipClient);
+  private static native void nativeStartSend(long nativeAndroidVoipClient);
+  private static native void nativeStopSend(long nativeAndroidVoipClient);
+  private static native void nativeStartPlayout(long nativeAndroidVoipClient);
+  private static native void nativeStopPlayout(long nativeAndroidVoipClient);
+  private static native void nativeDelete(long nativeAndroidVoipClient);
+}
diff --git a/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.cc b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.cc
new file mode 100644
index 0000000000..cf07e87e50
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.cc
@@ -0,0 +1,514 @@
+/*
+ *  Copyright 2020 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/androidvoip/jni/android_voip_client.h"
+
+#include <errno.h>
+#include <sys/socket.h>
+#include <algorithm>
+#include <map>
+#include <memory>
+#include <unordered_map>
+#include <unordered_set>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/voip/voip_codec.h"
+#include "api/voip/voip_engine_factory.h"
+#include "api/voip/voip_network.h"
+#include "examples/androidvoip/generated_jni/VoipClient_jni.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/network.h"
+#include "rtc_base/socket_server.h"
+#include "sdk/android/native_api/audio_device_module/audio_device_android.h"
+#include "sdk/android/native_api/jni/java_types.h"
+#include "sdk/android/native_api/jni/jvm.h"
+#include "sdk/android/native_api/jni/scoped_java_ref.h"
+
+namespace {
+
+#define RUN_ON_VOIP_THREAD(method, ...)                              \
+  if (!voip_thread_->IsCurrent()) {                                  \
+    voip_thread_->PostTask(                                          \
+        std::bind(&AndroidVoipClient::method, this, ##__VA_ARGS__)); \
+    return;                                                          \
+  }                                                                  \
+  RTC_DCHECK_RUN_ON(voip_thread_.get());
+
+// Connects a UDP socket to a public address and returns the local
+// address associated with it. Since it binds to the "any" address
+// internally, it returns the default local address on a multi-homed
+// endpoint. Implementation copied from
+// BasicNetworkManager::QueryDefaultLocalAddress.
+rtc::IPAddress QueryDefaultLocalAddress(int family) {
+  const char kPublicIPv4Host[] = "8.8.8.8";
+  const char kPublicIPv6Host[] = "2001:4860:4860::8888";
+  const int kPublicPort = 53;
+  std::unique_ptr<rtc::Thread> thread = rtc::Thread::CreateWithSocketServer();
+
+  RTC_DCHECK(thread->socketserver() != nullptr);
+  RTC_DCHECK(family == AF_INET || family == AF_INET6);
+
+  std::unique_ptr<rtc::Socket> socket(
+      thread->socketserver()->CreateSocket(family, SOCK_DGRAM));
+  if (!socket) {
+    RTC_LOG_ERR(LS_ERROR) << "Socket creation failed";
+    return rtc::IPAddress();
+  }
+
+  auto host = family == AF_INET ? kPublicIPv4Host : kPublicIPv6Host;
+  if (socket->Connect(rtc::SocketAddress(host, kPublicPort)) < 0) {
+    if (socket->GetError() != ENETUNREACH &&
+        socket->GetError() != EHOSTUNREACH) {
+      RTC_LOG(LS_INFO) << "Connect failed with " << socket->GetError();
+    }
+    return rtc::IPAddress();
+  }
+  return socket->GetLocalAddress().ipaddr();
+}
+
+// Assigned payload type for supported built-in codecs. PCMU, PCMA,
+// and G722 have set payload types. Whereas opus, ISAC, and ILBC
+// have dynamic payload types.
+enum class PayloadType : int {
+  kPcmu = 0,
+  kPcma = 8,
+  kG722 = 9,
+  kOpus = 96,
+  kIsac = 97,
+  kIlbc = 98,
+};
+
+// Returns the payload type corresponding to codec_name. Only
+// supports the built-in codecs.
+int GetPayloadType(const std::string& codec_name) {
+  RTC_DCHECK(codec_name == "PCMU" || codec_name == "PCMA" ||
+             codec_name == "G722" || codec_name == "opus" ||
+             codec_name == "ISAC" || codec_name == "ILBC");
+
+  if (codec_name == "PCMU") {
+    return static_cast<int>(PayloadType::kPcmu);
+  } else if (codec_name == "PCMA") {
+    return static_cast<int>(PayloadType::kPcma);
+  } else if (codec_name == "G722") {
+    return static_cast<int>(PayloadType::kG722);
+  } else if (codec_name == "opus") {
+    return static_cast<int>(PayloadType::kOpus);
+  } else if (codec_name == "ISAC") {
+    return static_cast<int>(PayloadType::kIsac);
+  } else if (codec_name == "ILBC") {
+    return static_cast<int>(PayloadType::kIlbc);
+  }
+
+  RTC_DCHECK_NOTREACHED();
+  return -1;
+}
+
+}  // namespace
+
+namespace webrtc_examples {
+
+void AndroidVoipClient::Init(
+    JNIEnv* env,
+    const webrtc::JavaParamRef<jobject>& application_context) {
+  webrtc::VoipEngineConfig config;
+  config.encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
+  config.decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
+  config.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
+  config.audio_device_module =
+      webrtc::CreateJavaAudioDeviceModule(env, application_context.obj());
+  config.audio_processing = webrtc::AudioProcessingBuilder().Create();
+
+  voip_thread_->Start();
+
+  // Due to consistent thread requirement on
+  // modules/audio_device/android/audio_device_template.h,
+  // code is invoked in the context of voip_thread_.
+  voip_thread_->BlockingCall([this, &config] {
+    RTC_DCHECK_RUN_ON(voip_thread_.get());
+
+    supported_codecs_ = config.encoder_factory->GetSupportedEncoders();
+    env_ = webrtc::AttachCurrentThreadIfNeeded();
+    voip_engine_ = webrtc::CreateVoipEngine(std::move(config));
+  });
+}
+
+AndroidVoipClient::~AndroidVoipClient() {
+  voip_thread_->BlockingCall([this] {
+    RTC_DCHECK_RUN_ON(voip_thread_.get());
+
+    JavaVM* jvm = nullptr;
+    env_->GetJavaVM(&jvm);
+    if (!jvm) {
+      RTC_LOG(LS_ERROR) << "Failed to retrieve JVM";
+      return;
+    }
+    jint res = jvm->DetachCurrentThread();
+    if (res != JNI_OK) {
+      RTC_LOG(LS_ERROR) << "DetachCurrentThread failed: " << res;
+    }
+  });
+
+  voip_thread_->Stop();
+}
+
+AndroidVoipClient* AndroidVoipClient::Create(
+    JNIEnv* env,
+    const webrtc::JavaParamRef<jobject>& application_context,
+    const webrtc::JavaParamRef<jobject>& j_voip_client) {
+  // Using `new` to access a non-public constructor.
+  auto voip_client =
+      absl::WrapUnique(new AndroidVoipClient(env, j_voip_client));
+  voip_client->Init(env, application_context);
+  return voip_client.release();
+}
+
+void AndroidVoipClient::GetSupportedCodecs(JNIEnv* env) {
+  RUN_ON_VOIP_THREAD(GetSupportedCodecs, env);
+
+  std::vector<std::string> names;
+  for (const webrtc::AudioCodecSpec& spec : supported_codecs_) {
+    names.push_back(spec.format.name);
+  }
+  webrtc::ScopedJavaLocalRef<jstring> (*convert_function)(
+      JNIEnv*, const std::string&) = &webrtc::NativeToJavaString;
+  Java_VoipClient_onGetSupportedCodecsCompleted(
+      env_, j_voip_client_, NativeToJavaList(env_, names, convert_function));
+}
+
+void AndroidVoipClient::GetLocalIPAddress(JNIEnv* env) {
+  RUN_ON_VOIP_THREAD(GetLocalIPAddress, env);
+
+  std::string local_ip_address;
+  rtc::IPAddress ipv4_address = QueryDefaultLocalAddress(AF_INET);
+  if (!ipv4_address.IsNil()) {
+    local_ip_address = ipv4_address.ToString();
+  } else {
+    rtc::IPAddress ipv6_address = QueryDefaultLocalAddress(AF_INET6);
+    if (!ipv6_address.IsNil()) {
+      local_ip_address = ipv6_address.ToString();
+    }
+  }
+  Java_VoipClient_onGetLocalIPAddressCompleted(
+      env_, j_voip_client_, webrtc::NativeToJavaString(env_, local_ip_address));
+}
+
+void AndroidVoipClient::SetEncoder(const std::string& encoder) {
+  RTC_DCHECK_RUN_ON(voip_thread_.get());
+
+  if (!channel_) {
+    RTC_LOG(LS_ERROR) << "Channel has not been created";
+    return;
+  }
+  for (const webrtc::AudioCodecSpec& codec : supported_codecs_) {
+    if (codec.format.name == encoder) {
+      webrtc::VoipResult result = voip_engine_->Codec().SetSendCodec(
+          *channel_, GetPayloadType(codec.format.name), codec.format);
+      RTC_CHECK(result == webrtc::VoipResult::kOk);
+      return;
+    }
+  }
+}
+
+void AndroidVoipClient::SetEncoder(
+    JNIEnv* env,
+    const webrtc::JavaParamRef<jstring>& j_encoder_string) {
+  const std::string& chosen_encoder =
+      webrtc::JavaToNativeString(env, j_encoder_string);
+  voip_thread_->PostTask(
+      [this, chosen_encoder] { SetEncoder(chosen_encoder); });
+}
+
+void AndroidVoipClient::SetDecoders(const std::vector<std::string>& decoders) {
+  RTC_DCHECK_RUN_ON(voip_thread_.get());
+
+  if (!channel_) {
+    RTC_LOG(LS_ERROR) << "Channel has not been created";
+    return;
+  }
+  std::map<int, webrtc::SdpAudioFormat> decoder_specs;
+  for (const webrtc::AudioCodecSpec& codec : supported_codecs_) {
+    if (std::find(decoders.begin(), decoders.end(), codec.format.name) !=
+        decoders.end()) {
+      decoder_specs.insert({GetPayloadType(codec.format.name), codec.format});
+    }
+  }
+
+  webrtc::VoipResult result =
+      voip_engine_->Codec().SetReceiveCodecs(*channel_, decoder_specs);
+  RTC_CHECK(result == webrtc::VoipResult::kOk);
+}
+
+void AndroidVoipClient::SetDecoders(
+    JNIEnv* env,
+    const webrtc::JavaParamRef<jobject>& j_decoder_strings) {
+  const std::vector<std::string>& chosen_decoders =
+      webrtc::JavaListToNativeVector<std::string, jstring>(
+          env, j_decoder_strings, &webrtc::JavaToNativeString);
+  voip_thread_->PostTask(
+      [this, chosen_decoders] { SetDecoders(chosen_decoders); });
+}
+
+void AndroidVoipClient::SetLocalAddress(const std::string& ip_address,
+                                        const int port_number) {
+  RTC_DCHECK_RUN_ON(voip_thread_.get());
+
+  rtp_local_address_ = rtc::SocketAddress(ip_address, port_number);
+  rtcp_local_address_ = rtc::SocketAddress(ip_address, port_number + 1);
+}
+
+void AndroidVoipClient::SetLocalAddress(
+    JNIEnv* env,
+    const webrtc::JavaParamRef<jstring>& j_ip_address_string,
+    jint j_port_number_int) {
+  const std::string& ip_address =
+      webrtc::JavaToNativeString(env, j_ip_address_string);
+  voip_thread_->PostTask([this, ip_address, j_port_number_int] {
+    SetLocalAddress(ip_address, j_port_number_int);
+  });
+}
+
+void AndroidVoipClient::SetRemoteAddress(const std::string& ip_address,
+                                         const int port_number) {
+  RTC_DCHECK_RUN_ON(voip_thread_.get());
+
+  rtp_remote_address_ = rtc::SocketAddress(ip_address, port_number);
+  rtcp_remote_address_ = rtc::SocketAddress(ip_address, port_number + 1);
+}
+
+void AndroidVoipClient::SetRemoteAddress(
+    JNIEnv* env,
+    const webrtc::JavaParamRef<jstring>& j_ip_address_string,
+    jint j_port_number_int) {
+  const std::string& ip_address =
+      webrtc::JavaToNativeString(env, j_ip_address_string);
+  voip_thread_->PostTask([this, ip_address, j_port_number_int] {
+    SetRemoteAddress(ip_address, j_port_number_int);
+  });
+}
+
+void AndroidVoipClient::StartSession(JNIEnv* env) {
+  RUN_ON_VOIP_THREAD(StartSession, env);
+
+  // CreateChannel guarantees to return valid channel id.
+  channel_ = voip_engine_->Base().CreateChannel(this, absl::nullopt);
+
+  rtp_socket_.reset(rtc::AsyncUDPSocket::Create(voip_thread_->socketserver(),
+                                                rtp_local_address_));
+  if (!rtp_socket_) {
+    RTC_LOG_ERR(LS_ERROR) << "Socket creation failed";
+    Java_VoipClient_onStartSessionCompleted(env_, j_voip_client_,
+                                            /*isSuccessful=*/false);
+    return;
+  }
+  rtp_socket_->SignalReadPacket.connect(
+      this, &AndroidVoipClient::OnSignalReadRTPPacket);
+
+  rtcp_socket_.reset(rtc::AsyncUDPSocket::Create(voip_thread_->socketserver(),
+                                                 rtcp_local_address_));
+  if (!rtcp_socket_) {
+    RTC_LOG_ERR(LS_ERROR) << "Socket creation failed";
+    Java_VoipClient_onStartSessionCompleted(env_, j_voip_client_,
+                                            /*isSuccessful=*/false);
+    return;
+  }
+  rtcp_socket_->SignalReadPacket.connect(
+      this, &AndroidVoipClient::OnSignalReadRTCPPacket);
+  Java_VoipClient_onStartSessionCompleted(env_, j_voip_client_,
+                                          /*isSuccessful=*/true);
+}
+
+void AndroidVoipClient::StopSession(JNIEnv* env) {
+  RUN_ON_VOIP_THREAD(StopSession, env);
+
+  if (!channel_) {
+    RTC_LOG(LS_ERROR) << "Channel has not been created";
+    Java_VoipClient_onStopSessionCompleted(env_, j_voip_client_,
+                                           /*isSuccessful=*/false);
+    return;
+  }
+  if (voip_engine_->Base().StopSend(*channel_) != webrtc::VoipResult::kOk ||
+      voip_engine_->Base().StopPlayout(*channel_) != webrtc::VoipResult::kOk) {
+    Java_VoipClient_onStopSessionCompleted(env_, j_voip_client_,
+                                           /*isSuccessful=*/false);
+    return;
+  }
+
+  rtp_socket_->Close();
+  rtcp_socket_->Close();
+
+  webrtc::VoipResult result = voip_engine_->Base().ReleaseChannel(*channel_);
+  RTC_CHECK(result == webrtc::VoipResult::kOk);
+
+  channel_ = absl::nullopt;
+  Java_VoipClient_onStopSessionCompleted(env_, j_voip_client_,
+                                         /*isSuccessful=*/true);
+}
+
+void AndroidVoipClient::StartSend(JNIEnv* env) {
+  RUN_ON_VOIP_THREAD(StartSend, env);
+
+  if (!channel_) {
+    RTC_LOG(LS_ERROR) << "Channel has not been created";
+    Java_VoipClient_onStartSendCompleted(env_, j_voip_client_,
+                                         /*isSuccessful=*/false);
+    return;
+  }
+  bool sending_started =
+      (voip_engine_->Base().StartSend(*channel_) == webrtc::VoipResult::kOk);
+  Java_VoipClient_onStartSendCompleted(env_, j_voip_client_, sending_started);
+}
+
+void AndroidVoipClient::StopSend(JNIEnv* env) {
+  RUN_ON_VOIP_THREAD(StopSend, env);
+
+  if (!channel_) {
+    RTC_LOG(LS_ERROR) << "Channel has not been created";
+    Java_VoipClient_onStopSendCompleted(env_, j_voip_client_,
+                                        /*isSuccessful=*/false);
+    return;
+  }
+  bool sending_stopped =
+      (voip_engine_->Base().StopSend(*channel_) == webrtc::VoipResult::kOk);
+  Java_VoipClient_onStopSendCompleted(env_, j_voip_client_, sending_stopped);
+}
+
+void AndroidVoipClient::StartPlayout(JNIEnv* env) {
+  RUN_ON_VOIP_THREAD(StartPlayout, env);
+
+  if (!channel_) {
+    RTC_LOG(LS_ERROR) << "Channel has not been created";
+    Java_VoipClient_onStartPlayoutCompleted(env_, j_voip_client_,
+                                            /*isSuccessful=*/false);
+    return;
+  }
+  bool playout_started =
+      (voip_engine_->Base().StartPlayout(*channel_) == webrtc::VoipResult::kOk);
+  Java_VoipClient_onStartPlayoutCompleted(env_, j_voip_client_,
+                                          playout_started);
+}
+
+void AndroidVoipClient::StopPlayout(JNIEnv* env) {
+  RUN_ON_VOIP_THREAD(StopPlayout, env);
+
+  if (!channel_) {
+    RTC_LOG(LS_ERROR) << "Channel has not been created";
+    Java_VoipClient_onStopPlayoutCompleted(env_, j_voip_client_,
+                                           /*isSuccessful=*/false);
+    return;
+  }
+  bool playout_stopped =
+      (voip_engine_->Base().StopPlayout(*channel_) == webrtc::VoipResult::kOk);
+  Java_VoipClient_onStopPlayoutCompleted(env_, j_voip_client_, playout_stopped);
+}
+
+void AndroidVoipClient::Delete(JNIEnv* env) {
+  delete this;
+}
+
+void AndroidVoipClient::SendRtpPacket(const std::vector<uint8_t>& packet_copy) {
+  RTC_DCHECK_RUN_ON(voip_thread_.get());
+
+  if (!rtp_socket_->SendTo(packet_copy.data(), packet_copy.size(),
+                           rtp_remote_address_, rtc::PacketOptions())) {
+    RTC_LOG(LS_ERROR) << "Failed to send RTP packet";
+  }
+}
+
+bool AndroidVoipClient::SendRtp(const uint8_t* packet,
+                                size_t length,
+                                const webrtc::PacketOptions& options) {
+  std::vector<uint8_t> packet_copy(packet, packet + length);
+  voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] {
+    SendRtpPacket(packet_copy);
+  });
+  return true;
+}
+
+void AndroidVoipClient::SendRtcpPacket(
+    const std::vector<uint8_t>& packet_copy) {
+  RTC_DCHECK_RUN_ON(voip_thread_.get());
+
+  if (!rtcp_socket_->SendTo(packet_copy.data(), packet_copy.size(),
+                            rtcp_remote_address_, rtc::PacketOptions())) {
+    RTC_LOG(LS_ERROR) << "Failed to send RTCP packet";
+  }
+}
+
+bool AndroidVoipClient::SendRtcp(const uint8_t* packet, size_t length) {
+  std::vector<uint8_t> packet_copy(packet, packet + length);
+  voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] {
+    SendRtcpPacket(packet_copy);
+  });
+  return true;
+}
+
+void AndroidVoipClient::ReadRTPPacket(const std::vector<uint8_t>& packet_copy) {
+  RTC_DCHECK_RUN_ON(voip_thread_.get());
+
+  if (!channel_) {
+    RTC_LOG(LS_ERROR) << "Channel has not been created";
+    return;
+  }
+  webrtc::VoipResult result = voip_engine_->Network().ReceivedRTPPacket(
+      *channel_,
+      rtc::ArrayView<const uint8_t>(packet_copy.data(), packet_copy.size()));
+  RTC_CHECK(result == webrtc::VoipResult::kOk);
+}
+
+void AndroidVoipClient::OnSignalReadRTPPacket(rtc::AsyncPacketSocket* socket,
+                                              const char* rtp_packet,
+                                              size_t size,
+                                              const rtc::SocketAddress& addr,
+                                              const int64_t& timestamp) {
+  std::vector<uint8_t> packet_copy(rtp_packet, rtp_packet + size);
+  voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] {
+    ReadRTPPacket(packet_copy);
+  });
+}
+
+void AndroidVoipClient::ReadRTCPPacket(
+    const std::vector<uint8_t>& packet_copy) {
+  RTC_DCHECK_RUN_ON(voip_thread_.get());
+
+  if (!channel_) {
+    RTC_LOG(LS_ERROR) << "Channel has not been created";
+    return;
+  }
+  webrtc::VoipResult result = voip_engine_->Network().ReceivedRTCPPacket(
+      *channel_,
+      rtc::ArrayView<const uint8_t>(packet_copy.data(), packet_copy.size()));
+  RTC_CHECK(result == webrtc::VoipResult::kOk);
+}
+
+void AndroidVoipClient::OnSignalReadRTCPPacket(rtc::AsyncPacketSocket* socket,
+                                               const char* rtcp_packet,
+                                               size_t size,
+                                               const rtc::SocketAddress& addr,
+                                               const int64_t& timestamp) {
+  std::vector<uint8_t> packet_copy(rtcp_packet, rtcp_packet + size);
+  voip_thread_->PostTask([this, packet_copy = std::move(packet_copy)] {
+    ReadRTCPPacket(packet_copy);
+  });
+}
+
+static jlong JNI_VoipClient_CreateClient(
+    JNIEnv* env,
+    const webrtc::JavaParamRef<jobject>& application_context,
+    const webrtc::JavaParamRef<jobject>& j_voip_client) {
+  return webrtc::NativeToJavaPointer(
+      AndroidVoipClient::Create(env, application_context, j_voip_client));
+}
+
+}  // namespace webrtc_examples
diff --git a/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.h b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.h
new file mode 100644
index 0000000000..8e1edd5ef9
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidvoip/jni/android_voip_client.h
@@ -0,0 +1,189 @@
+/*
+ *  Copyright 2020 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_ANDROIDVOIP_JNI_ANDROID_VOIP_CLIENT_H_
+#define EXAMPLES_ANDROIDVOIP_JNI_ANDROID_VOIP_CLIENT_H_
+
+#include <jni.h>
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/audio_codecs/audio_format.h"
+#include "api/call/transport.h"
+#include "api/voip/voip_base.h"
+#include "api/voip/voip_engine.h"
+#include "rtc_base/async_packet_socket.h"
+#include "rtc_base/async_udp_socket.h"
+#include "rtc_base/socket_address.h"
+#include "rtc_base/third_party/sigslot/sigslot.h"
+#include "rtc_base/thread.h"
+#include "sdk/android/native_api/jni/scoped_java_ref.h"
+
+namespace webrtc_examples {
+
+// AndroidVoipClient facilitates the use of the VoIP API defined in
+// api/voip/voip_engine.h. One instance of AndroidVoipClient should
+// suffice for most VoIP applications. AndroidVoipClient implements
+// webrtc::Transport to send RTP/RTCP packets to the remote endpoint.
+// It also creates methods (slots) for sockets to connect to in
+// order to receive RTP/RTCP packets. AndroidVoipClient does all
+// operations with rtc::Thread (voip_thread_), this is to comply
+// with consistent thread usage requirement with ProcessThread used
+// within VoipEngine, as well as providing asynchronicity to the
+// caller. AndroidVoipClient is meant to be used by Java through JNI.
+class AndroidVoipClient : public webrtc::Transport,
+                          public sigslot::has_slots<> {
+ public:
+  // Returns a pointer to an AndroidVoipClient object. Clients should
+  // use this factory method to create AndroidVoipClient objects. The
+  // method will return a nullptr in case of initialization errors.
+  // It is the client's responsibility to delete the pointer when
+  // they are done with it (this class provides a Delete() method).
+  static AndroidVoipClient* Create(
+      JNIEnv* env,
+      const webrtc::JavaParamRef<jobject>& application_context,
+      const webrtc::JavaParamRef<jobject>& j_voip_client);
+
+  ~AndroidVoipClient() override;
+
+  // Provides client with a Java List of Strings containing names of
+  // the built-in supported codecs through callback.
+  void GetSupportedCodecs(JNIEnv* env);
+
+  // Provides client with a Java String of the default local IPv4 address
+  // through callback. If IPv4 address is not found, provide the default
+  // local IPv6 address. If IPv6 address is not found, provide an empty
+  // string.
+  void GetLocalIPAddress(JNIEnv* env);
+
+  // Sets the encoder used by the VoIP API.
+  void SetEncoder(JNIEnv* env,
+                  const webrtc::JavaParamRef<jstring>& j_encoder_string);
+
+  // Sets the decoders used by the VoIP API.
+  void SetDecoders(JNIEnv* env,
+                   const webrtc::JavaParamRef<jobject>& j_decoder_strings);
+
+  // Sets two local/remote addresses, one for RTP packets, and another for
+  // RTCP packets. The RTP address will have IP address j_ip_address_string
+  // and port number j_port_number_int, the RTCP address will have IP address
+  // j_ip_address_string and port number j_port_number_int+1.
+  void SetLocalAddress(JNIEnv* env,
+                       const webrtc::JavaParamRef<jstring>& j_ip_address_string,
+                       jint j_port_number_int);
+  void SetRemoteAddress(
+      JNIEnv* env,
+      const webrtc::JavaParamRef<jstring>& j_ip_address_string,
+      jint j_port_number_int);
+
+  // Starts a VoIP session, then calls a callback method with a boolean
+  // value indicating if the session has started successfully. The VoIP
+  // operations below can only be used after a session has already started.
+  void StartSession(JNIEnv* env);
+
+  // Stops the current session, then calls a callback method with a
+  // boolean value indicating if the session has stopped successfully.
+  void StopSession(JNIEnv* env);
+
+  // Starts sending RTP/RTCP packets to the remote endpoint, then calls
+  // a callback method with a boolean value indicating if sending
+  // has started successfully.
+  void StartSend(JNIEnv* env);
+
+  // Stops sending RTP/RTCP packets to the remote endpoint, then calls
+  // a callback method with a boolean value indicating if sending
+  // has stopped successfully.
+  void StopSend(JNIEnv* env);
+
+  // Starts playing out the voice data received from the remote endpoint,
+  // then calls a callback method with a boolean value indicating if
+  // playout has started successfully.
+  void StartPlayout(JNIEnv* env);
+
+  // Stops playing out the voice data received from the remote endpoint,
+  // then calls a callback method with a boolean value indicating if
+  // playout has stopped successfully.
+  void StopPlayout(JNIEnv* env);
+
+  // Deletes this object. Used by client when they are done.
+  void Delete(JNIEnv* env);
+
+  // Implementation for Transport.
+  bool SendRtp(const uint8_t* packet,
+               size_t length,
+               const webrtc::PacketOptions& options) override;
+  bool SendRtcp(const uint8_t* packet, size_t length) override;
+
+  // Slots for sockets to connect to.
+  void OnSignalReadRTPPacket(rtc::AsyncPacketSocket* socket,
+                             const char* rtp_packet,
+                             size_t size,
+                             const rtc::SocketAddress& addr,
+                             const int64_t& timestamp);
+  void OnSignalReadRTCPPacket(rtc::AsyncPacketSocket* socket,
+                              const char* rtcp_packet,
+                              size_t size,
+                              const rtc::SocketAddress& addr,
+                              const int64_t& timestamp);
+
+ private:
+  AndroidVoipClient(JNIEnv* env,
+                    const webrtc::JavaParamRef<jobject>& j_voip_client)
+      : voip_thread_(rtc::Thread::CreateWithSocketServer()),
+        j_voip_client_(env, j_voip_client) {}
+
+  void Init(JNIEnv* env,
+            const webrtc::JavaParamRef<jobject>& application_context);
+
+  // Overloaded methods having native C++ variables as arguments.
+  void SetEncoder(const std::string& encoder);
+  void SetDecoders(const std::vector<std::string>& decoders);
+  void SetLocalAddress(const std::string& ip_address, int port_number);
+  void SetRemoteAddress(const std::string& ip_address, int port_number);
+
+  // Methods to send and receive RTP/RTCP packets. Takes in a
+  // copy of a packet as a vector to prolong the lifetime of
+  // the packet as these methods will be called asynchronously.
+  void SendRtpPacket(const std::vector<uint8_t>& packet_copy);
+  void SendRtcpPacket(const std::vector<uint8_t>& packet_copy);
+  void ReadRTPPacket(const std::vector<uint8_t>& packet_copy);
+  void ReadRTCPPacket(const std::vector<uint8_t>& packet_copy);
+
+  // Used to invoke operations and send/receive RTP/RTCP packets.
+  std::unique_ptr<rtc::Thread> voip_thread_;
+  // Reference to the VoipClient java instance used to
+  // invoke callbacks when operations are finished.
+  webrtc::ScopedJavaGlobalRef<jobject> j_voip_client_
+      RTC_GUARDED_BY(voip_thread_);
+  // A list of AudioCodecSpec supported by the built-in
+  // encoder/decoder factories.
+  std::vector<webrtc::AudioCodecSpec> supported_codecs_
+      RTC_GUARDED_BY(voip_thread_);
+  // A JNI context used by the voip_thread_.
+  JNIEnv* env_ RTC_GUARDED_BY(voip_thread_);
+  // The entry point to all VoIP APIs.
+  std::unique_ptr<webrtc::VoipEngine> voip_engine_ RTC_GUARDED_BY(voip_thread_);
+  // Used by the VoIP API to facilitate a VoIP session.
+  absl::optional<webrtc::ChannelId> channel_ RTC_GUARDED_BY(voip_thread_);
+  // Members below are used for network related operations.
+  std::unique_ptr<rtc::AsyncUDPSocket> rtp_socket_ RTC_GUARDED_BY(voip_thread_);
+  std::unique_ptr<rtc::AsyncUDPSocket> rtcp_socket_
+      RTC_GUARDED_BY(voip_thread_);
+  rtc::SocketAddress rtp_local_address_ RTC_GUARDED_BY(voip_thread_);
+  rtc::SocketAddress rtcp_local_address_ RTC_GUARDED_BY(voip_thread_);
+  rtc::SocketAddress rtp_remote_address_ RTC_GUARDED_BY(voip_thread_);
+  rtc::SocketAddress rtcp_remote_address_ RTC_GUARDED_BY(voip_thread_);
+};
+
+}  // namespace webrtc_examples
+
+#endif  // EXAMPLES_ANDROIDVOIP_JNI_ANDROID_VOIP_CLIENT_H_
diff --git a/third_party/libwebrtc/examples/androidvoip/jni/onload.cc b/third_party/libwebrtc/examples/androidvoip/jni/onload.cc
new file mode 100644
index 0000000000..b952de348b
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidvoip/jni/onload.cc
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <jni.h>
+
+#include "rtc_base/ssl_adapter.h"
+#include "sdk/android/native_api/base/init.h"
+
+namespace webrtc_examples {
+
+extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM* jvm, void* reserved) {
+  webrtc::InitAndroid(jvm);
+  RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
+  return JNI_VERSION_1_6;
+}
+
+extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM* jvm, void* reserved) {
+  RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()";
+}
+
+}  // namespace webrtc_examples
diff --git a/third_party/libwebrtc/examples/androidvoip/res/layout/activity_main.xml b/third_party/libwebrtc/examples/androidvoip/res/layout/activity_main.xml
new file mode 100644
index 0000000000..c7fa5a9b31
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidvoip/res/layout/activity_main.xml
@@ -0,0 +1,303 @@
+<?xml version="1.0" encoding="utf-8"?>
+<ScrollView
+  xmlns:android="http://schemas.android.com/apk/res/android"
+  xmlns:app="http://schemas.android.com/apk/res-auto"
+  xmlns:tools="http://schemas.android.com/tools"
+  android:id="@+id/scroll_view"
+  android:layout_width="match_parent"
+  android:layout_height="wrap_content"
+  android:focusable="true"
+  android:focusableInTouchMode="true"
+  tools:context="org.webrtc.examples.androidvoip.MainActivity">
+
+  <LinearLayout
+    android:orientation="vertical"
+    android:layout_width="match_parent"
+    android:layout_height="match_parent"
+    android:padding="8dp">
+
+    <TextView
+      android:layout_width="match_parent"
+      android:layout_height="wrap_content"
+      android:layout_marginBottom="15dp"
+      android:layout_marginLeft="15dp"
+      android:layout_marginTop="15dp"
+      android:text="@string/local_endpoint_text_view"
+      android:textSize="19dp"
+      android:textStyle="bold"
+      android:textColor="@color/almost_black" />
+
+    <!--Local IP Adress-->
+    <LinearLayout
+      android:orientation="horizontal"
+      android:layout_width="match_parent"
+      android:layout_height="48dp"
+      android:layout_gravity="center_vertical" >
+
+      <TextView
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_marginTop="12dp"
+        android:layout_marginLeft="15dp"
+        android:layout_marginRight="15dp"
+        android:text="@string/ip_address_text_view"
+        android:textSize="16dp" />
+
+      <TextView
+        android:id="@+id/local_ip_address_text_view"
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_marginTop="12dp"
+        android:layout_marginRight="15dp"
+        android:textSize="16dp" />
+
+    </LinearLayout>
+
+    <!--Local Port Number-->
+    <LinearLayout
+      android:orientation="horizontal"
+      android:layout_width="match_parent"
+      android:layout_height="48dp"
+      android:layout_gravity="center_vertical">
+
+      <TextView
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_marginBottom="15dp"
+        android:layout_marginLeft="15dp"
+        android:layout_marginRight="15dp"
+        android:text="@string/port_number_text_view"
+        android:textSize="16dp" />
+
+      <EditText
+        android:id="@+id/local_port_number_edit_text"
+        android:layout_width="0dp"
+        android:layout_height="match_parent"
+        android:layout_marginRight="15dp"
+        android:layout_weight="1"
+        android:text="10000"
+        android:inputType="number"
+        android:textSize="16dp" />
+
+    </LinearLayout>
+
+    <TextView
+      android:layout_width="match_parent"
+      android:layout_height="wrap_content"
+      android:layout_marginBottom="15dp"
+      android:layout_marginLeft="15dp"
+      android:layout_marginTop="30dp"
+      android:text="@string/remote_endpoint_text_view"
+      android:textSize="19dp"
+      android:textStyle="bold"
+      android:textColor="@color/almost_black" />
+
+    <!--Remote IP Adress-->
+    <LinearLayout
+      android:orientation="horizontal"
+      android:layout_width="match_parent"
+      android:layout_height="48dp"
+      android:layout_gravity="center_vertical">
+
+      <TextView
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_marginBottom="15dp"
+        android:layout_marginLeft="15dp"
+        android:layout_marginRight="15dp"
+        android:text="@string/ip_address_text_view"
+        android:textSize="16dp" />
+
+      <EditText
+        android:id="@+id/remote_ip_address_edit_text"
+        android:layout_width="0dp"
+        android:layout_height="wrap_content"
+        android:layout_marginRight="15dp"
+        android:layout_weight="1"
+        android:inputType="number"
+        android:digits="0123456789."
+        android:textSize="16dp" />
+
+    </LinearLayout>
+
+    <!--Remote Port Number-->
+    <LinearLayout
+      android:orientation="horizontal"
+      android:layout_width="match_parent"
+      android:layout_height="48dp"
+      android:layout_gravity="center_vertical">
+
+      <TextView
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_marginBottom="15dp"
+        android:layout_marginLeft="15dp"
+        android:layout_marginRight="15dp"
+        android:text="@string/port_number_text_view"
+        android:textSize="16dp" />
+
+      <EditText
+        android:id="@+id/remote_port_number_edit_text"
+        android:layout_width="0dp"
+        android:layout_height="match_parent"
+        android:layout_marginRight="15dp"
+        android:layout_weight="1"
+        android:text="10000"
+        android:inputType="number"
+        android:textSize="16dp" />
+
+    </LinearLayout>
+
+    <TextView
+      android:layout_width="wrap_content"
+      android:layout_height="wrap_content"
+      android:layout_marginBottom="15dp"
+      android:layout_marginLeft="15dp"
+      android:layout_marginTop="30dp"
+      android:text="@string/encoder_text_view"
+      android:textSize="19dp"
+      android:textStyle="bold"
+      android:textColor="@color/almost_black" />
+
+    <Spinner
+      android:id="@+id/encoder_spinner"
+      android:layout_width="wrap_content"
+      android:layout_height="wrap_content"
+      android:layout_marginBottom="15dp"
+      android:layout_marginLeft="15dp"
+      android:layout_marginTop="10dp"/>
+
+    <LinearLayout
+      android:orientation="horizontal"
+      android:layout_width="match_parent"
+      android:layout_height="48dp"
+      android:layout_marginTop="20dp"
+      android:layout_gravity="center_vertical">
+
+      <TextView
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_marginBottom="15dp"
+        android:layout_marginLeft="15dp"
+        android:layout_marginRight="25dp"
+        android:text="@string/decoder_text_view"
+        android:textSize="19dp"
+        android:textStyle="bold"
+        android:textColor="@color/almost_black" />
+
+      <Button
+        android:id="@+id/decoder_selection_button"
+        android:text="@string/decoder_selection_button"
+        style="?android:attr/buttonBarButtonStyle"
+        android:layout_width="0dp"
+        android:layout_height="match_parent"
+        android:layout_marginRight="15dp"
+        android:layout_weight="1" />
+
+    </LinearLayout>
+
+
+    <TextView
+      android:id="@+id/decoders_text_view"
+      android:layout_width="wrap_content"
+      android:layout_height="wrap_content"
+      android:layout_marginTop="15dp"
+      android:layout_marginBottom="30dp"
+      android:layout_marginLeft="15dp"
+      android:layout_marginRight="15dp"
+      android:text="@string/decoders_text_view_default"
+      android:textSize="16dp" />
+
+
+    <RelativeLayout
+      android:id="@+id/switch_layout"
+      android:layout_width="fill_parent"
+      android:layout_height="wrap_content"
+      android:layout_marginTop="15dp"
+      android:visibility="gone" >
+
+      <View
+        android:id="@+id/divider"
+        android:layout_width="match_parent"
+        android:layout_height="1dp"
+        android:layout_marginLeft="15dp"
+        android:layout_marginRight="15dp"
+        android:layout_marginBottom="45dp"
+        android:background="@color/light_gray" />
+
+      <LinearLayout
+        android:id="@+id/start_send_switch_layout"
+        android:orientation="horizontal"
+        android:layout_width="match_parent"
+        android:layout_height="48dp"
+        android:layout_gravity="center_vertical"
+        android:layout_below="@id/divider" >
+
+        <TextView
+          android:layout_width="wrap_content"
+          android:layout_height="match_parent"
+          android:layout_marginLeft="15dp"
+          android:gravity="left"
+          android:layout_weight="1"
+          android:text="@string/start_send_text_view"
+          android:textSize="16dp" />
+
+        <Switch
+          android:id="@+id/start_send_switch"
+          android:layout_width="wrap_content"
+          android:layout_height="match_parent"
+          android:layout_marginRight="15dp"
+          android:gravity="right"
+          android:layout_weight="1" />
+
+      </LinearLayout>
+
+      <LinearLayout
+        android:orientation="horizontal"
+        android:layout_width="match_parent"
+        android:layout_height="48dp"
+        android:layout_gravity="center_vertical"
+        android:layout_below="@id/start_send_switch_layout">
+
+        <TextView
+          android:id="@+id/start_playout_text_view"
+          android:layout_width="wrap_content"
+          android:layout_height="match_parent"
+          android:layout_marginLeft="15dp"
+          android:gravity="left"
+          android:layout_weight="1"
+          android:text="@string/start_playout_text_view"
+          android:textSize="16dp" />
+
+        <Switch
+          android:id="@+id/start_playout_switch"
+          android:layout_width="wrap_content"
+          android:layout_height="match_parent"
+          android:layout_marginRight="15dp"
+          android:gravity="right"
+          android:layout_weight="1" />
+
+      </LinearLayout>
+
+    </RelativeLayout>
+
+    <LinearLayout
+      android:layout_width="match_parent"
+      android:layout_height="match_parent"
+      android:gravity="center"
+      android:orientation="vertical" >
+
+      <ToggleButton
+        android:id="@+id/session_button"
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:layout_margin="8dp"
+        android:textOff="@string/session_button_text_off"
+        android:textOn="@string/session_button_text_on"
+        style="?android:attr/buttonStyle" />
+
+    </LinearLayout>
+
+  </LinearLayout>
+
+</ScrollView>
diff --git a/third_party/libwebrtc/examples/androidvoip/res/values/colors.xml b/third_party/libwebrtc/examples/androidvoip/res/values/colors.xml
new file mode 100644
index 0000000000..4dadaa9941
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidvoip/res/values/colors.xml
@@ -0,0 +1,5 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+   <color name="almost_black">#484848</color>
+   <color name="light_gray">#D3D3D3</color>
+</resources>
\ No newline at end of file
diff --git a/third_party/libwebrtc/examples/androidvoip/res/values/strings.xml b/third_party/libwebrtc/examples/androidvoip/res/values/strings.xml
new file mode 100644
index 0000000000..d519bfbbb6
--- /dev/null
+++ b/third_party/libwebrtc/examples/androidvoip/res/values/strings.xml
@@ -0,0 +1,19 @@
+<resources>
+  <string name="app_name">androidvoip</string>
+  <string name="local_endpoint_text_view">Local Endpoint</string>
+  <string name="remote_endpoint_text_view">Remote Endpoint</string>
+  <string name="ip_address_text_view">IP Address:</string>
+  <string name="port_number_text_view">Port Number:</string>
+  <string name="encoder_text_view">Select Encoder</string>
+  <string name="decoder_text_view">Select Decoder</string>
+  <string name="decoder_selection_button">Configure Selection</string>
+  <string name="decoders_text_view_default">No decoders selected</string>
+  <string name="dialog_title">Choose Decoders</string>
+  <string name="ok_label">Ok</string>
+  <string name="dismiss_label">Dismiss</string>
+  <string name="clear_all_label">Clear All</string>
+  <string name="start_send_text_view">Start Sending</string>
+  <string name="start_playout_text_view">Start Playout</string>
+  <string name="session_button_text_off">Start Session</string>
+  <string name="session_button_text_on">Stop Session</string>
+</resources>
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient+Internal.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient+Internal.h
new file mode 100644
index 0000000000..31e0e4dd7c
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient+Internal.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDAppClient.h"
+
+#import "sdk/objc/api/peerconnection/RTCPeerConnection.h"
+
+#import "ARDRoomServerClient.h"
+#import "ARDSignalingChannel.h"
+#import "ARDTURNClient.h"
+
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+
+@interface ARDAppClient () <ARDSignalingChannelDelegate, RTC_OBJC_TYPE (RTCPeerConnectionDelegate)>
+
+// All properties should only be mutated from the main queue.
+@property(nonatomic, strong) id<ARDRoomServerClient> roomServerClient;
+@property(nonatomic, strong) id<ARDSignalingChannel> channel;
+@property(nonatomic, strong) id<ARDSignalingChannel> loopbackChannel;
+@property(nonatomic, strong) id<ARDTURNClient> turnClient;
+
+@property(nonatomic, strong) RTC_OBJC_TYPE(RTCPeerConnection) * peerConnection;
+@property(nonatomic, strong) RTC_OBJC_TYPE(RTCPeerConnectionFactory) * factory;
+@property(nonatomic, strong) NSMutableArray *messageQueue;
+
+@property(nonatomic, assign) BOOL isTurnComplete;
+@property(nonatomic, assign) BOOL hasReceivedSdp;
+@property(nonatomic, readonly) BOOL hasJoinedRoomServerRoom;
+
+@property(nonatomic, strong) NSString *roomId;
+@property(nonatomic, strong) NSString *clientId;
+@property(nonatomic, assign) BOOL isInitiator;
+@property(nonatomic, strong) NSMutableArray *iceServers;
+@property(nonatomic, strong) NSURL *webSocketURL;
+@property(nonatomic, strong) NSURL *webSocketRestURL;
+@property(nonatomic, readonly) BOOL isLoopback;
+
+@property(nonatomic, strong) RTC_OBJC_TYPE(RTCMediaConstraints) * defaultPeerConnectionConstraints;
+
+- (instancetype)initWithRoomServerClient:(id<ARDRoomServerClient>)rsClient
+                        signalingChannel:(id<ARDSignalingChannel>)channel
+                              turnClient:(id<ARDTURNClient>)turnClient
+                                delegate:(id<ARDAppClientDelegate>)delegate;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient.h
new file mode 100644
index 0000000000..91d2cef1ce
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient.h
@@ -0,0 +1,87 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "sdk/objc/api/peerconnection/RTCPeerConnection.h"
+#import "sdk/objc/api/peerconnection/RTCVideoTrack.h"
+
+typedef NS_ENUM(NSInteger, ARDAppClientState) {
+  // Disconnected from servers.
+  kARDAppClientStateDisconnected,
+  // Connecting to servers.
+  kARDAppClientStateConnecting,
+  // Connected to servers.
+  kARDAppClientStateConnected,
+};
+
+@class ARDAppClient;
+@class ARDSettingsModel;
+@class ARDExternalSampleCapturer;
+@class RTC_OBJC_TYPE(RTCMediaConstraints);
+@class RTC_OBJC_TYPE(RTCCameraVideoCapturer);
+@class RTC_OBJC_TYPE(RTCFileVideoCapturer);
+
+// The delegate is informed of pertinent events and will be called on the
+// main queue.
+@protocol ARDAppClientDelegate <NSObject>
+
+- (void)appClient:(ARDAppClient *)client didChangeState:(ARDAppClientState)state;
+
+- (void)appClient:(ARDAppClient *)client didChangeConnectionState:(RTCIceConnectionState)state;
+
+- (void)appClient:(ARDAppClient *)client
+    didCreateLocalCapturer:(RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)localCapturer;
+
+- (void)appClient:(ARDAppClient *)client
+    didReceiveLocalVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)localVideoTrack;
+
+- (void)appClient:(ARDAppClient *)client
+    didReceiveRemoteVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)remoteVideoTrack;
+
+- (void)appClient:(ARDAppClient *)client didError:(NSError *)error;
+
+- (void)appClient:(ARDAppClient *)client didGetStats:(RTC_OBJC_TYPE(RTCStatisticsReport) *)stats;
+
+@optional
+- (void)appClient:(ARDAppClient *)client
+    didCreateLocalFileCapturer:(RTC_OBJC_TYPE(RTCFileVideoCapturer) *)fileCapturer;
+
+- (void)appClient:(ARDAppClient *)client
+    didCreateLocalExternalSampleCapturer:(ARDExternalSampleCapturer *)externalSampleCapturer;
+
+@end
+
+// Handles connections to the AppRTC server for a given room. Methods on this
+// class should only be called from the main queue.
+@interface ARDAppClient : NSObject
+
+// If `shouldGetStats` is true, stats will be reported in 1s intervals through
+// the delegate.
+@property(nonatomic, assign) BOOL shouldGetStats;
+@property(nonatomic, readonly) ARDAppClientState state;
+@property(nonatomic, weak) id<ARDAppClientDelegate> delegate;
+@property(nonatomic, assign, getter=isBroadcast) BOOL broadcast;
+
+// Convenience constructor since all expected use cases will need a delegate
+// in order to receive remote tracks.
+- (instancetype)initWithDelegate:(id<ARDAppClientDelegate>)delegate;
+
+// Establishes a connection with the AppRTC servers for the given room id.
+// `settings` is an object containing settings such as video codec for the call.
+// If `isLoopback` is true, the call will connect to itself.
+- (void)connectToRoomWithId:(NSString *)roomId
+                   settings:(ARDSettingsModel *)settings
+                 isLoopback:(BOOL)isLoopback;
+
+// Disconnects from the AppRTC servers and any connected clients.
+- (void)disconnect;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient.m
new file mode 100644
index 0000000000..4420972598
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppClient.m
@@ -0,0 +1,899 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDAppClient+Internal.h"
+
+#import "sdk/objc/api/peerconnection/RTCAudioTrack.h"
+#import "sdk/objc/api/peerconnection/RTCConfiguration.h"
+#import "sdk/objc/api/peerconnection/RTCFileLogger.h"
+#import "sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent.h"
+#import "sdk/objc/api/peerconnection/RTCIceServer.h"
+#import "sdk/objc/api/peerconnection/RTCMediaConstraints.h"
+#import "sdk/objc/api/peerconnection/RTCMediaStream.h"
+#import "sdk/objc/api/peerconnection/RTCPeerConnectionFactory.h"
+#import "sdk/objc/api/peerconnection/RTCRtpSender.h"
+#import "sdk/objc/api/peerconnection/RTCRtpTransceiver.h"
+#import "sdk/objc/api/peerconnection/RTCTracing.h"
+#import "sdk/objc/api/peerconnection/RTCVideoSource.h"
+#import "sdk/objc/api/peerconnection/RTCVideoTrack.h"
+#import "sdk/objc/base/RTCLogging.h"
+#import "sdk/objc/components/capturer/RTCCameraVideoCapturer.h"
+#import "sdk/objc/components/capturer/RTCFileVideoCapturer.h"
+#import "sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h"
+#import "sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h"
+
+#import "ARDAppEngineClient.h"
+#import "ARDExternalSampleCapturer.h"
+#import "ARDJoinResponse.h"
+#import "ARDMessageResponse.h"
+#import "ARDSettingsModel.h"
+#import "ARDSignalingMessage.h"
+#import "ARDTURNClient+Internal.h"
+#import "ARDUtilities.h"
+#import "ARDWebSocketChannel.h"
+#import "RTCIceCandidate+JSON.h"
+#import "RTCSessionDescription+JSON.h"
+
+static NSString * const kARDIceServerRequestUrl = @"https://appr.tc/params";
+
+static NSString * const kARDAppClientErrorDomain = @"ARDAppClient";
+static NSInteger const kARDAppClientErrorUnknown = -1;
+static NSInteger const kARDAppClientErrorRoomFull = -2;
+static NSInteger const kARDAppClientErrorCreateSDP = -3;
+static NSInteger const kARDAppClientErrorSetSDP = -4;
+static NSInteger const kARDAppClientErrorInvalidClient = -5;
+static NSInteger const kARDAppClientErrorInvalidRoom = -6;
+static NSString * const kARDMediaStreamId = @"ARDAMS";
+static NSString * const kARDAudioTrackId = @"ARDAMSa0";
+static NSString * const kARDVideoTrackId = @"ARDAMSv0";
+static NSString * const kARDVideoTrackKind = @"video";
+
+// TODO(tkchin): Add these as UI options.
+#if defined(WEBRTC_IOS)
+static BOOL const kARDAppClientEnableTracing = NO;
+static BOOL const kARDAppClientEnableRtcEventLog = YES;
+static int64_t const kARDAppClientAecDumpMaxSizeInBytes = 5e6;  // 5 MB.
+static int64_t const kARDAppClientRtcEventLogMaxSizeInBytes = 5e6;  // 5 MB.
+#endif
+static int const kKbpsMultiplier = 1000;
+
+// We need a proxy to NSTimer because it causes a strong retain cycle. When
+// using the proxy, `invalidate` must be called before it properly deallocs.
+@interface ARDTimerProxy : NSObject
+
+- (instancetype)initWithInterval:(NSTimeInterval)interval
+                         repeats:(BOOL)repeats
+                    timerHandler:(void (^)(void))timerHandler;
+- (void)invalidate;
+
+@end
+
+@implementation ARDTimerProxy {
+  NSTimer *_timer;
+  void (^_timerHandler)(void);
+}
+
+- (instancetype)initWithInterval:(NSTimeInterval)interval
+                         repeats:(BOOL)repeats
+                    timerHandler:(void (^)(void))timerHandler {
+  NSParameterAssert(timerHandler);
+  if (self = [super init]) {
+    _timerHandler = timerHandler;
+    _timer = [NSTimer scheduledTimerWithTimeInterval:interval
+                                              target:self
+                                            selector:@selector(timerDidFire:)
+                                            userInfo:nil
+                                             repeats:repeats];
+  }
+  return self;
+}
+
+- (void)invalidate {
+  [_timer invalidate];
+}
+
+- (void)timerDidFire:(NSTimer *)timer {
+  _timerHandler();
+}
+
+@end
+
+@implementation ARDAppClient {
+  RTC_OBJC_TYPE(RTCFileLogger) * _fileLogger;
+  ARDTimerProxy *_statsTimer;
+  ARDSettingsModel *_settings;
+  RTC_OBJC_TYPE(RTCVideoTrack) * _localVideoTrack;
+}
+
+@synthesize shouldGetStats = _shouldGetStats;
+@synthesize state = _state;
+@synthesize delegate = _delegate;
+@synthesize roomServerClient = _roomServerClient;
+@synthesize channel = _channel;
+@synthesize loopbackChannel = _loopbackChannel;
+@synthesize turnClient = _turnClient;
+@synthesize peerConnection = _peerConnection;
+@synthesize factory = _factory;
+@synthesize messageQueue = _messageQueue;
+@synthesize isTurnComplete = _isTurnComplete;
+@synthesize hasReceivedSdp  = _hasReceivedSdp;
+@synthesize roomId = _roomId;
+@synthesize clientId = _clientId;
+@synthesize isInitiator = _isInitiator;
+@synthesize iceServers = _iceServers;
+@synthesize webSocketURL = _websocketURL;
+@synthesize webSocketRestURL = _websocketRestURL;
+@synthesize defaultPeerConnectionConstraints =
+    _defaultPeerConnectionConstraints;
+@synthesize isLoopback = _isLoopback;
+@synthesize broadcast = _broadcast;
+
+- (instancetype)init {
+  return [self initWithDelegate:nil];
+}
+
+- (instancetype)initWithDelegate:(id<ARDAppClientDelegate>)delegate {
+  if (self = [super init]) {
+    _roomServerClient = [[ARDAppEngineClient alloc] init];
+    _delegate = delegate;
+    NSURL *turnRequestURL = [NSURL URLWithString:kARDIceServerRequestUrl];
+    _turnClient = [[ARDTURNClient alloc] initWithURL:turnRequestURL];
+    [self configure];
+  }
+  return self;
+}
+
+// TODO(tkchin): Provide signaling channel factory interface so we can recreate
+// channel if we need to on network failure. Also, make this the default public
+// constructor.
+- (instancetype)initWithRoomServerClient:(id<ARDRoomServerClient>)rsClient
+                        signalingChannel:(id<ARDSignalingChannel>)channel
+                              turnClient:(id<ARDTURNClient>)turnClient
+                                delegate:(id<ARDAppClientDelegate>)delegate {
+  NSParameterAssert(rsClient);
+  NSParameterAssert(channel);
+  NSParameterAssert(turnClient);
+  if (self = [super init]) {
+    _roomServerClient = rsClient;
+    _channel = channel;
+    _turnClient = turnClient;
+    _delegate = delegate;
+    [self configure];
+  }
+  return self;
+}
+
+- (void)configure {
+  _messageQueue = [NSMutableArray array];
+  _iceServers = [NSMutableArray array];
+  _fileLogger = [[RTC_OBJC_TYPE(RTCFileLogger) alloc] init];
+  [_fileLogger start];
+}
+
+- (void)dealloc {
+  self.shouldGetStats = NO;
+  [self disconnect];
+}
+
+- (void)setShouldGetStats:(BOOL)shouldGetStats {
+  if (_shouldGetStats == shouldGetStats) {
+    return;
+  }
+  if (shouldGetStats) {
+    __weak ARDAppClient *weakSelf = self;
+    _statsTimer = [[ARDTimerProxy alloc] initWithInterval:1
+                                                  repeats:YES
+                                             timerHandler:^{
+      ARDAppClient *strongSelf = weakSelf;
+      [strongSelf.peerConnection statisticsWithCompletionHandler:^(
+                                     RTC_OBJC_TYPE(RTCStatisticsReport) * stats) {
+        dispatch_async(dispatch_get_main_queue(), ^{
+          ARDAppClient *strongSelf = weakSelf;
+          [strongSelf.delegate appClient:strongSelf didGetStats:stats];
+        });
+      }];
+    }];
+  } else {
+    [_statsTimer invalidate];
+    _statsTimer = nil;
+  }
+  _shouldGetStats = shouldGetStats;
+}
+
+- (void)setState:(ARDAppClientState)state {
+  if (_state == state) {
+    return;
+  }
+  _state = state;
+  [_delegate appClient:self didChangeState:_state];
+}
+
+- (void)connectToRoomWithId:(NSString *)roomId
+                   settings:(ARDSettingsModel *)settings
+                 isLoopback:(BOOL)isLoopback {
+  NSParameterAssert(roomId.length);
+  NSParameterAssert(_state == kARDAppClientStateDisconnected);
+  _settings = settings;
+  _isLoopback = isLoopback;
+  self.state = kARDAppClientStateConnecting;
+
+  RTC_OBJC_TYPE(RTCDefaultVideoDecoderFactory) *decoderFactory =
+      [[RTC_OBJC_TYPE(RTCDefaultVideoDecoderFactory) alloc] init];
+  RTC_OBJC_TYPE(RTCDefaultVideoEncoderFactory) *encoderFactory =
+      [[RTC_OBJC_TYPE(RTCDefaultVideoEncoderFactory) alloc] init];
+  encoderFactory.preferredCodec = [settings currentVideoCodecSettingFromStore];
+  _factory =
+      [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] initWithEncoderFactory:encoderFactory
+                                                               decoderFactory:decoderFactory];
+
+#if defined(WEBRTC_IOS)
+  if (kARDAppClientEnableTracing) {
+    NSString *filePath = [self documentsFilePathForFileName:@"webrtc-trace.txt"];
+    RTCStartInternalCapture(filePath);
+  }
+#endif
+
+  // Request TURN.
+  __weak ARDAppClient *weakSelf = self;
+  [_turnClient requestServersWithCompletionHandler:^(NSArray *turnServers,
+                                                     NSError *error) {
+    if (error) {
+      RTCLogError(@"Error retrieving TURN servers: %@", error.localizedDescription);
+    }
+    ARDAppClient *strongSelf = weakSelf;
+    [strongSelf.iceServers addObjectsFromArray:turnServers];
+    strongSelf.isTurnComplete = YES;
+    [strongSelf startSignalingIfReady];
+  }];
+
+  // Join room on room server.
+  [_roomServerClient joinRoomWithRoomId:roomId
+                             isLoopback:isLoopback
+      completionHandler:^(ARDJoinResponse *response, NSError *error) {
+    ARDAppClient *strongSelf = weakSelf;
+    if (error) {
+      [strongSelf.delegate appClient:strongSelf didError:error];
+      return;
+    }
+    NSError *joinError =
+        [[strongSelf class] errorForJoinResultType:response.result];
+    if (joinError) {
+      RTCLogError(@"Failed to join room:%@ on room server.", roomId);
+      [strongSelf disconnect];
+      [strongSelf.delegate appClient:strongSelf didError:joinError];
+      return;
+    }
+    RTCLog(@"Joined room:%@ on room server.", roomId);
+    strongSelf.roomId = response.roomId;
+    strongSelf.clientId = response.clientId;
+    strongSelf.isInitiator = response.isInitiator;
+    for (ARDSignalingMessage *message in response.messages) {
+      if (message.type == kARDSignalingMessageTypeOffer ||
+          message.type == kARDSignalingMessageTypeAnswer) {
+        strongSelf.hasReceivedSdp = YES;
+        [strongSelf.messageQueue insertObject:message atIndex:0];
+      } else {
+        [strongSelf.messageQueue addObject:message];
+      }
+    }
+    strongSelf.webSocketURL = response.webSocketURL;
+    strongSelf.webSocketRestURL = response.webSocketRestURL;
+    [strongSelf registerWithColliderIfReady];
+    [strongSelf startSignalingIfReady];
+  }];
+}
+
+- (void)disconnect {
+  if (_state == kARDAppClientStateDisconnected) {
+    return;
+  }
+  if (self.hasJoinedRoomServerRoom) {
+    [_roomServerClient leaveRoomWithRoomId:_roomId
+                                  clientId:_clientId
+                         completionHandler:nil];
+  }
+  if (_channel) {
+    if (_channel.state == kARDSignalingChannelStateRegistered) {
+      // Tell the other client we're hanging up.
+      ARDByeMessage *byeMessage = [[ARDByeMessage alloc] init];
+      [_channel sendMessage:byeMessage];
+    }
+    // Disconnect from collider.
+    _channel = nil;
+  }
+  _clientId = nil;
+  _roomId = nil;
+  _isInitiator = NO;
+  _hasReceivedSdp = NO;
+  _messageQueue = [NSMutableArray array];
+  _localVideoTrack = nil;
+#if defined(WEBRTC_IOS)
+  [_factory stopAecDump];
+  [_peerConnection stopRtcEventLog];
+#endif
+  [_peerConnection close];
+  _peerConnection = nil;
+  self.state = kARDAppClientStateDisconnected;
+#if defined(WEBRTC_IOS)
+  if (kARDAppClientEnableTracing) {
+    RTCStopInternalCapture();
+  }
+#endif
+}
+
+#pragma mark - ARDSignalingChannelDelegate
+
+- (void)channel:(id<ARDSignalingChannel>)channel
+    didReceiveMessage:(ARDSignalingMessage *)message {
+  switch (message.type) {
+    case kARDSignalingMessageTypeOffer:
+    case kARDSignalingMessageTypeAnswer:
+      // Offers and answers must be processed before any other message, so we
+      // place them at the front of the queue.
+      _hasReceivedSdp = YES;
+      [_messageQueue insertObject:message atIndex:0];
+      break;
+    case kARDSignalingMessageTypeCandidate:
+    case kARDSignalingMessageTypeCandidateRemoval:
+      [_messageQueue addObject:message];
+      break;
+    case kARDSignalingMessageTypeBye:
+      // Disconnects can be processed immediately.
+      [self processSignalingMessage:message];
+      return;
+  }
+  [self drainMessageQueueIfReady];
+}
+
+- (void)channel:(id<ARDSignalingChannel>)channel
+    didChangeState:(ARDSignalingChannelState)state {
+  switch (state) {
+    case kARDSignalingChannelStateOpen:
+      break;
+    case kARDSignalingChannelStateRegistered:
+      break;
+    case kARDSignalingChannelStateClosed:
+    case kARDSignalingChannelStateError:
+      // TODO(tkchin): reconnection scenarios. Right now we just disconnect
+      // completely if the websocket connection fails.
+      [self disconnect];
+      break;
+  }
+}
+
+#pragma mark - RTC_OBJC_TYPE(RTCPeerConnectionDelegate)
+// Callbacks for this delegate occur on non-main thread and need to be
+// dispatched back to main queue as needed.
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeSignalingState:(RTCSignalingState)stateChanged {
+  RTCLog(@"Signaling state changed: %ld", (long)stateChanged);
+}
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+          didAddStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream {
+  RTCLog(@"Stream with %lu video tracks and %lu audio tracks was added.",
+         (unsigned long)stream.videoTracks.count,
+         (unsigned long)stream.audioTracks.count);
+}
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didStartReceivingOnTransceiver:(RTC_OBJC_TYPE(RTCRtpTransceiver) *)transceiver {
+  RTC_OBJC_TYPE(RTCMediaStreamTrack) *track = transceiver.receiver.track;
+  RTCLog(@"Now receiving %@ on track %@.", track.kind, track.trackId);
+}
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+       didRemoveStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream {
+  RTCLog(@"Stream was removed.");
+}
+
+- (void)peerConnectionShouldNegotiate:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection {
+  RTCLog(@"WARNING: Renegotiation needed but unimplemented.");
+}
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeIceConnectionState:(RTCIceConnectionState)newState {
+  RTCLog(@"ICE state changed: %ld", (long)newState);
+  dispatch_async(dispatch_get_main_queue(), ^{
+    [self.delegate appClient:self didChangeConnectionState:newState];
+  });
+}
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeConnectionState:(RTCPeerConnectionState)newState {
+  RTCLog(@"ICE+DTLS state changed: %ld", (long)newState);
+}
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeIceGatheringState:(RTCIceGatheringState)newState {
+  RTCLog(@"ICE gathering state changed: %ld", (long)newState);
+}
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didGenerateIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate {
+  dispatch_async(dispatch_get_main_queue(), ^{
+    ARDICECandidateMessage *message =
+        [[ARDICECandidateMessage alloc] initWithCandidate:candidate];
+    [self sendSignalingMessage:message];
+  });
+}
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didFailToGatherIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) *)event {
+  RTCLog(@"Failed to gather ICE candidate. address: %@, port: %d, url: %@, errorCode: %d, "
+         @"errorText: %@",
+         event.address,
+         event.port,
+         event.url,
+         event.errorCode,
+         event.errorText);
+}
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didRemoveIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates {
+  dispatch_async(dispatch_get_main_queue(), ^{
+    ARDICECandidateRemovalMessage *message =
+        [[ARDICECandidateRemovalMessage alloc]
+            initWithRemovedCandidates:candidates];
+    [self sendSignalingMessage:message];
+  });
+}
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+     didChangeLocalCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)local
+    didChangeRemoteCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)remote
+              lastReceivedMs:(int)lastDataReceivedMs
+               didHaveReason:(NSString *)reason {
+  RTCLog(@"ICE candidate pair changed because: %@", reason);
+}
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didOpenDataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel {
+}
+
+#pragma mark - RTCSessionDescriptionDelegate
+// Callbacks for this delegate occur on non-main thread and need to be
+// dispatched back to main queue as needed.
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didCreateSessionDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
+                          error:(NSError *)error {
+  dispatch_async(dispatch_get_main_queue(), ^{
+    if (error) {
+      RTCLogError(@"Failed to create session description. Error: %@", error);
+      [self disconnect];
+      NSDictionary *userInfo = @{
+        NSLocalizedDescriptionKey: @"Failed to create session description.",
+      };
+      NSError *sdpError =
+          [[NSError alloc] initWithDomain:kARDAppClientErrorDomain
+                                     code:kARDAppClientErrorCreateSDP
+                                 userInfo:userInfo];
+      [self.delegate appClient:self didError:sdpError];
+      return;
+    }
+    __weak ARDAppClient *weakSelf = self;
+    [self.peerConnection setLocalDescription:sdp
+                           completionHandler:^(NSError *error) {
+                             ARDAppClient *strongSelf = weakSelf;
+                             [strongSelf peerConnection:strongSelf.peerConnection
+                                 didSetSessionDescriptionWithError:error];
+                           }];
+    ARDSessionDescriptionMessage *message =
+        [[ARDSessionDescriptionMessage alloc] initWithDescription:sdp];
+    [self sendSignalingMessage:message];
+    [self setMaxBitrateForPeerConnectionVideoSender];
+  });
+}
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didSetSessionDescriptionWithError:(NSError *)error {
+  dispatch_async(dispatch_get_main_queue(), ^{
+    if (error) {
+      RTCLogError(@"Failed to set session description. Error: %@", error);
+      [self disconnect];
+      NSDictionary *userInfo = @{
+        NSLocalizedDescriptionKey: @"Failed to set session description.",
+      };
+      NSError *sdpError =
+          [[NSError alloc] initWithDomain:kARDAppClientErrorDomain
+                                     code:kARDAppClientErrorSetSDP
+                                 userInfo:userInfo];
+      [self.delegate appClient:self didError:sdpError];
+      return;
+    }
+    // If we're answering and we've just set the remote offer we need to create
+    // an answer and set the local description.
+    if (!self.isInitiator && !self.peerConnection.localDescription) {
+      RTC_OBJC_TYPE(RTCMediaConstraints) *constraints = [self defaultAnswerConstraints];
+      __weak ARDAppClient *weakSelf = self;
+      [self.peerConnection
+          answerForConstraints:constraints
+             completionHandler:^(RTC_OBJC_TYPE(RTCSessionDescription) * sdp, NSError * error) {
+               ARDAppClient *strongSelf = weakSelf;
+               [strongSelf peerConnection:strongSelf.peerConnection
+                   didCreateSessionDescription:sdp
+                                         error:error];
+             }];
+    }
+  });
+}
+
+#pragma mark - Private
+
+#if defined(WEBRTC_IOS)
+
+- (NSString *)documentsFilePathForFileName:(NSString *)fileName {
+  NSParameterAssert(fileName.length);
+  NSArray *paths = NSSearchPathForDirectoriesInDomains(
+      NSDocumentDirectory, NSUserDomainMask, YES);
+  NSString *documentsDirPath = paths.firstObject;
+  NSString *filePath =
+      [documentsDirPath stringByAppendingPathComponent:fileName];
+  return filePath;
+}
+
+#endif
+
+- (BOOL)hasJoinedRoomServerRoom {
+  return _clientId.length;
+}
+
+// Begins the peer connection connection process if we have both joined a room
+// on the room server and tried to obtain a TURN server. Otherwise does nothing.
+// A peer connection object will be created with a stream that contains local
+// audio and video capture. If this client is the caller, an offer is created as
+// well, otherwise the client will wait for an offer to arrive.
+- (void)startSignalingIfReady {
+  if (!_isTurnComplete || !self.hasJoinedRoomServerRoom) {
+    return;
+  }
+  self.state = kARDAppClientStateConnected;
+
+  // Create peer connection.
+  RTC_OBJC_TYPE(RTCMediaConstraints) *constraints = [self defaultPeerConnectionConstraints];
+  RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+  RTC_OBJC_TYPE(RTCCertificate) *pcert = [RTC_OBJC_TYPE(RTCCertificate)
+      generateCertificateWithParams:@{@"expires" : @100000, @"name" : @"RSASSA-PKCS1-v1_5"}];
+  config.iceServers = _iceServers;
+  config.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
+  config.certificate = pcert;
+
+  _peerConnection = [_factory peerConnectionWithConfiguration:config
+                                                  constraints:constraints
+                                                     delegate:self];
+  // Create AV senders.
+  [self createMediaSenders];
+  if (_isInitiator) {
+    // Send offer.
+    __weak ARDAppClient *weakSelf = self;
+    [_peerConnection
+        offerForConstraints:[self defaultOfferConstraints]
+          completionHandler:^(RTC_OBJC_TYPE(RTCSessionDescription) * sdp, NSError * error) {
+            ARDAppClient *strongSelf = weakSelf;
+            [strongSelf peerConnection:strongSelf.peerConnection
+                didCreateSessionDescription:sdp
+                                      error:error];
+          }];
+  } else {
+    // Check if we've received an offer.
+    [self drainMessageQueueIfReady];
+  }
+#if defined(WEBRTC_IOS)
+  // Start event log.
+  if (kARDAppClientEnableRtcEventLog) {
+    NSString *filePath = [self documentsFilePathForFileName:@"webrtc-rtceventlog"];
+    if (![_peerConnection startRtcEventLogWithFilePath:filePath
+                                 maxSizeInBytes:kARDAppClientRtcEventLogMaxSizeInBytes]) {
+      RTCLogError(@"Failed to start event logging.");
+    }
+  }
+
+  // Start aecdump diagnostic recording.
+  if ([_settings currentCreateAecDumpSettingFromStore]) {
+    NSString *filePath = [self documentsFilePathForFileName:@"webrtc-audio.aecdump"];
+    if (![_factory startAecDumpWithFilePath:filePath
+                             maxSizeInBytes:kARDAppClientAecDumpMaxSizeInBytes]) {
+      RTCLogError(@"Failed to start aec dump.");
+    }
+  }
+#endif
+}
+
+// Processes the messages that we've received from the room server and the
+// signaling channel. The offer or answer message must be processed before other
+// signaling messages, however they can arrive out of order. Hence, this method
+// only processes pending messages if there is a peer connection object and
+// if we have received either an offer or answer.
+- (void)drainMessageQueueIfReady {
+  if (!_peerConnection || !_hasReceivedSdp) {
+    return;
+  }
+  for (ARDSignalingMessage *message in _messageQueue) {
+    [self processSignalingMessage:message];
+  }
+  [_messageQueue removeAllObjects];
+}
+
+// Processes the given signaling message based on its type.
+- (void)processSignalingMessage:(ARDSignalingMessage *)message {
+  NSParameterAssert(_peerConnection ||
+      message.type == kARDSignalingMessageTypeBye);
+  switch (message.type) {
+    case kARDSignalingMessageTypeOffer:
+    case kARDSignalingMessageTypeAnswer: {
+      ARDSessionDescriptionMessage *sdpMessage =
+          (ARDSessionDescriptionMessage *)message;
+      RTC_OBJC_TYPE(RTCSessionDescription) *description = sdpMessage.sessionDescription;
+      __weak ARDAppClient *weakSelf = self;
+      [_peerConnection setRemoteDescription:description
+                          completionHandler:^(NSError *error) {
+                            ARDAppClient *strongSelf = weakSelf;
+                            [strongSelf peerConnection:strongSelf.peerConnection
+                                didSetSessionDescriptionWithError:error];
+                          }];
+      break;
+    }
+    case kARDSignalingMessageTypeCandidate: {
+      ARDICECandidateMessage *candidateMessage =
+          (ARDICECandidateMessage *)message;
+      __weak ARDAppClient *weakSelf = self;
+      [_peerConnection addIceCandidate:candidateMessage.candidate
+                     completionHandler:^(NSError *error) {
+                       ARDAppClient *strongSelf = weakSelf;
+                       if (error) {
+                         [strongSelf.delegate appClient:strongSelf didError:error];
+                       }
+                     }];
+      break;
+    }
+    case kARDSignalingMessageTypeCandidateRemoval: {
+      ARDICECandidateRemovalMessage *candidateMessage =
+          (ARDICECandidateRemovalMessage *)message;
+      [_peerConnection removeIceCandidates:candidateMessage.candidates];
+      break;
+    }
+    case kARDSignalingMessageTypeBye:
+      // Other client disconnected.
+      // TODO(tkchin): support waiting in room for next client. For now just
+      // disconnect.
+      [self disconnect];
+      break;
+  }
+}
+
+// Sends a signaling message to the other client. The caller will send messages
+// through the room server, whereas the callee will send messages over the
+// signaling channel.
+- (void)sendSignalingMessage:(ARDSignalingMessage *)message {
+  if (_isInitiator) {
+    __weak ARDAppClient *weakSelf = self;
+    [_roomServerClient sendMessage:message
+                         forRoomId:_roomId
+                          clientId:_clientId
+                 completionHandler:^(ARDMessageResponse *response,
+                                     NSError *error) {
+      ARDAppClient *strongSelf = weakSelf;
+      if (error) {
+        [strongSelf.delegate appClient:strongSelf didError:error];
+        return;
+      }
+      NSError *messageError =
+          [[strongSelf class] errorForMessageResultType:response.result];
+      if (messageError) {
+        [strongSelf.delegate appClient:strongSelf didError:messageError];
+        return;
+      }
+    }];
+  } else {
+    [_channel sendMessage:message];
+  }
+}
+
+- (void)setMaxBitrateForPeerConnectionVideoSender {
+  for (RTC_OBJC_TYPE(RTCRtpSender) * sender in _peerConnection.senders) {
+    if (sender.track != nil) {
+      if ([sender.track.kind isEqualToString:kARDVideoTrackKind]) {
+        [self setMaxBitrate:[_settings currentMaxBitrateSettingFromStore] forVideoSender:sender];
+      }
+    }
+  }
+}
+
+- (void)setMaxBitrate:(NSNumber *)maxBitrate forVideoSender:(RTC_OBJC_TYPE(RTCRtpSender) *)sender {
+  if (maxBitrate.intValue <= 0) {
+    return;
+  }
+
+  RTC_OBJC_TYPE(RTCRtpParameters) *parametersToModify = sender.parameters;
+  for (RTC_OBJC_TYPE(RTCRtpEncodingParameters) * encoding in parametersToModify.encodings) {
+    encoding.maxBitrateBps = @(maxBitrate.intValue * kKbpsMultiplier);
+  }
+  [sender setParameters:parametersToModify];
+}
+
+- (RTC_OBJC_TYPE(RTCRtpTransceiver) *)videoTransceiver {
+  for (RTC_OBJC_TYPE(RTCRtpTransceiver) * transceiver in _peerConnection.transceivers) {
+    if (transceiver.mediaType == RTCRtpMediaTypeVideo) {
+      return transceiver;
+    }
+  }
+  return nil;
+}
+
+- (void)createMediaSenders {
+  RTC_OBJC_TYPE(RTCMediaConstraints) *constraints = [self defaultMediaAudioConstraints];
+  RTC_OBJC_TYPE(RTCAudioSource) *source = [_factory audioSourceWithConstraints:constraints];
+  RTC_OBJC_TYPE(RTCAudioTrack) *track = [_factory audioTrackWithSource:source
+                                                               trackId:kARDAudioTrackId];
+  [_peerConnection addTrack:track streamIds:@[ kARDMediaStreamId ]];
+  _localVideoTrack = [self createLocalVideoTrack];
+  if (_localVideoTrack) {
+    [_peerConnection addTrack:_localVideoTrack streamIds:@[ kARDMediaStreamId ]];
+    [_delegate appClient:self didReceiveLocalVideoTrack:_localVideoTrack];
+    // We can set up rendering for the remote track right away since the transceiver already has an
+    // RTC_OBJC_TYPE(RTCRtpReceiver) with a track. The track will automatically get unmuted and
+    // produce frames once RTP is received.
+    RTC_OBJC_TYPE(RTCVideoTrack) *track =
+        (RTC_OBJC_TYPE(RTCVideoTrack) *)([self videoTransceiver].receiver.track);
+    [_delegate appClient:self didReceiveRemoteVideoTrack:track];
+  }
+}
+
+- (RTC_OBJC_TYPE(RTCVideoTrack) *)createLocalVideoTrack {
+  if ([_settings currentAudioOnlySettingFromStore]) {
+    return nil;
+  }
+
+  RTC_OBJC_TYPE(RTCVideoSource) *source = [_factory videoSource];
+
+#if !TARGET_IPHONE_SIMULATOR
+  if (self.isBroadcast) {
+    ARDExternalSampleCapturer *capturer =
+        [[ARDExternalSampleCapturer alloc] initWithDelegate:source];
+    [_delegate appClient:self didCreateLocalExternalSampleCapturer:capturer];
+  } else {
+    RTC_OBJC_TYPE(RTCCameraVideoCapturer) *capturer =
+        [[RTC_OBJC_TYPE(RTCCameraVideoCapturer) alloc] initWithDelegate:source];
+    [_delegate appClient:self didCreateLocalCapturer:capturer];
+  }
+#else
+#if defined(__IPHONE_11_0) && (__IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_11_0)
+  if (@available(iOS 10, *)) {
+    RTC_OBJC_TYPE(RTCFileVideoCapturer) *fileCapturer =
+        [[RTC_OBJC_TYPE(RTCFileVideoCapturer) alloc] initWithDelegate:source];
+    [_delegate appClient:self didCreateLocalFileCapturer:fileCapturer];
+  }
+#endif
+#endif
+
+  return [_factory videoTrackWithSource:source trackId:kARDVideoTrackId];
+}
+
+#pragma mark - Collider methods
+
+- (void)registerWithColliderIfReady {
+  if (!self.hasJoinedRoomServerRoom) {
+    return;
+  }
+  // Open WebSocket connection.
+  if (!_channel) {
+    _channel =
+        [[ARDWebSocketChannel alloc] initWithURL:_websocketURL
+                                         restURL:_websocketRestURL
+                                        delegate:self];
+    if (_isLoopback) {
+      _loopbackChannel =
+          [[ARDLoopbackWebSocketChannel alloc] initWithURL:_websocketURL
+                                                   restURL:_websocketRestURL];
+    }
+  }
+  [_channel registerForRoomId:_roomId clientId:_clientId];
+  if (_isLoopback) {
+    [_loopbackChannel registerForRoomId:_roomId clientId:@"LOOPBACK_CLIENT_ID"];
+  }
+}
+
+#pragma mark - Defaults
+
+- (RTC_OBJC_TYPE(RTCMediaConstraints) *)defaultMediaAudioConstraints {
+  NSDictionary *mandatoryConstraints = @{};
+  RTC_OBJC_TYPE(RTCMediaConstraints) *constraints =
+      [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:mandatoryConstraints
+                                                           optionalConstraints:nil];
+  return constraints;
+}
+
+- (RTC_OBJC_TYPE(RTCMediaConstraints) *)defaultAnswerConstraints {
+  return [self defaultOfferConstraints];
+}
+
+- (RTC_OBJC_TYPE(RTCMediaConstraints) *)defaultOfferConstraints {
+  NSDictionary *mandatoryConstraints = @{
+    @"OfferToReceiveAudio" : @"true",
+    @"OfferToReceiveVideo" : @"true"
+  };
+  RTC_OBJC_TYPE(RTCMediaConstraints) *constraints =
+      [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:mandatoryConstraints
+                                                           optionalConstraints:nil];
+  return constraints;
+}
+
+- (RTC_OBJC_TYPE(RTCMediaConstraints) *)defaultPeerConnectionConstraints {
+  if (_defaultPeerConnectionConstraints) {
+    return _defaultPeerConnectionConstraints;
+  }
+  NSString *value = _isLoopback ? @"false" : @"true";
+  NSDictionary *optionalConstraints = @{ @"DtlsSrtpKeyAgreement" : value };
+  RTC_OBJC_TYPE(RTCMediaConstraints) *constraints =
+      [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:nil
+                                                           optionalConstraints:optionalConstraints];
+  return constraints;
+}
+
+#pragma mark - Errors
+
++ (NSError *)errorForJoinResultType:(ARDJoinResultType)resultType {
+  NSError *error = nil;
+  switch (resultType) {
+    case kARDJoinResultTypeSuccess:
+      break;
+    case kARDJoinResultTypeUnknown: {
+      error = [[NSError alloc] initWithDomain:kARDAppClientErrorDomain
+                                         code:kARDAppClientErrorUnknown
+                                     userInfo:@{
+        NSLocalizedDescriptionKey: @"Unknown error.",
+      }];
+      break;
+    }
+    case kARDJoinResultTypeFull: {
+      error = [[NSError alloc] initWithDomain:kARDAppClientErrorDomain
+                                         code:kARDAppClientErrorRoomFull
+                                     userInfo:@{
+        NSLocalizedDescriptionKey: @"Room is full.",
+      }];
+      break;
+    }
+  }
+  return error;
+}
+
++ (NSError *)errorForMessageResultType:(ARDMessageResultType)resultType {
+  NSError *error = nil;
+  switch (resultType) {
+    case kARDMessageResultTypeSuccess:
+      break;
+    case kARDMessageResultTypeUnknown:
+      error = [[NSError alloc] initWithDomain:kARDAppClientErrorDomain
+                                         code:kARDAppClientErrorUnknown
+                                     userInfo:@{
+        NSLocalizedDescriptionKey: @"Unknown error.",
+      }];
+      break;
+    case kARDMessageResultTypeInvalidClient:
+      error = [[NSError alloc] initWithDomain:kARDAppClientErrorDomain
+                                         code:kARDAppClientErrorInvalidClient
+                                     userInfo:@{
+        NSLocalizedDescriptionKey: @"Invalid client.",
+      }];
+      break;
+    case kARDMessageResultTypeInvalidRoom:
+      error = [[NSError alloc] initWithDomain:kARDAppClientErrorDomain
+                                         code:kARDAppClientErrorInvalidRoom
+                                     userInfo:@{
+        NSLocalizedDescriptionKey: @"Invalid room.",
+      }];
+      break;
+  }
+  return error;
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.h
new file mode 100644
index 0000000000..7514f3645c
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.h
@@ -0,0 +1,14 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDRoomServerClient.h"
+
+@interface ARDAppEngineClient : NSObject <ARDRoomServerClient>
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.m
new file mode 100644
index 0000000000..5139de60d6
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.m
@@ -0,0 +1,175 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDAppEngineClient.h"
+
+#import "sdk/objc/base/RTCLogging.h"
+
+#import "ARDJoinResponse.h"
+#import "ARDMessageResponse.h"
+#import "ARDSignalingMessage.h"
+#import "ARDUtilities.h"
+
+// TODO(tkchin): move these to a configuration object.
+static NSString * const kARDRoomServerHostUrl =
+    @"https://appr.tc";
+static NSString * const kARDRoomServerJoinFormat =
+    @"https://appr.tc/join/%@";
+static NSString * const kARDRoomServerJoinFormatLoopback =
+    @"https://appr.tc/join/%@?debug=loopback";
+static NSString * const kARDRoomServerMessageFormat =
+    @"https://appr.tc/message/%@/%@";
+static NSString * const kARDRoomServerLeaveFormat =
+    @"https://appr.tc/leave/%@/%@";
+
+static NSString * const kARDAppEngineClientErrorDomain = @"ARDAppEngineClient";
+static NSInteger const kARDAppEngineClientErrorBadResponse = -1;
+
+@implementation ARDAppEngineClient
+
+#pragma mark - ARDRoomServerClient
+
+- (void)joinRoomWithRoomId:(NSString *)roomId
+                isLoopback:(BOOL)isLoopback
+         completionHandler:(void (^)(ARDJoinResponse *response,
+                                     NSError *error))completionHandler {
+  NSParameterAssert(roomId.length);
+
+  NSString *urlString = nil;
+  if (isLoopback) {
+    urlString =
+        [NSString stringWithFormat:kARDRoomServerJoinFormatLoopback, roomId];
+  } else {
+    urlString =
+        [NSString stringWithFormat:kARDRoomServerJoinFormat, roomId];
+  }
+
+  NSURL *roomURL = [NSURL URLWithString:urlString];
+  RTCLog(@"Joining room:%@ on room server.", roomId);
+  NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:roomURL];
+  request.HTTPMethod = @"POST";
+  [NSURLConnection sendAsyncRequest:request
+                  completionHandler:^(NSURLResponse *response, NSData *data, NSError *error) {
+                    if (error) {
+                      if (completionHandler) {
+                        completionHandler(nil, error);
+                      }
+                      return;
+                    }
+                    ARDJoinResponse *joinResponse = [ARDJoinResponse responseFromJSONData:data];
+                    if (!joinResponse) {
+                      if (completionHandler) {
+                        NSError *error = [[self class] badResponseError];
+                        completionHandler(nil, error);
+                      }
+                      return;
+                    }
+                    if (completionHandler) {
+                      completionHandler(joinResponse, nil);
+                    }
+                  }];
+}
+
+- (void)sendMessage:(ARDSignalingMessage *)message
+            forRoomId:(NSString *)roomId
+             clientId:(NSString *)clientId
+    completionHandler:(void (^)(ARDMessageResponse *response,
+                                NSError *error))completionHandler {
+  NSParameterAssert(message);
+  NSParameterAssert(roomId.length);
+  NSParameterAssert(clientId.length);
+
+  NSData *data = [message JSONData];
+  NSString *urlString =
+      [NSString stringWithFormat:
+          kARDRoomServerMessageFormat, roomId, clientId];
+  NSURL *url = [NSURL URLWithString:urlString];
+  RTCLog(@"C->RS POST: %@", message);
+  NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:url];
+  request.HTTPMethod = @"POST";
+  request.HTTPBody = data;
+  [NSURLConnection sendAsyncRequest:request
+                  completionHandler:^(NSURLResponse *response,
+                                      NSData *data,
+                                      NSError *error) {
+    if (error) {
+      if (completionHandler) {
+        completionHandler(nil, error);
+      }
+      return;
+    }
+    ARDMessageResponse *messageResponse =
+        [ARDMessageResponse responseFromJSONData:data];
+    if (!messageResponse) {
+      if (completionHandler) {
+        NSError *error = [[self class] badResponseError];
+        completionHandler(nil, error);
+      }
+      return;
+    }
+    if (completionHandler) {
+      completionHandler(messageResponse, nil);
+    }
+  }];
+}
+
+- (void)leaveRoomWithRoomId:(NSString *)roomId
+                   clientId:(NSString *)clientId
+          completionHandler:(void (^)(NSError *error))completionHandler {
+  NSParameterAssert(roomId.length);
+  NSParameterAssert(clientId.length);
+
+  NSString *urlString =
+      [NSString stringWithFormat:kARDRoomServerLeaveFormat, roomId, clientId];
+  NSURL *url = [NSURL URLWithString:urlString];
+  NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:url];
+  request.HTTPMethod = @"POST";
+
+  RTCLog(@"C->RS: BYE");
+  __block NSError *error = nil;
+
+  // We want a synchronous request so that we know that we've left the room on
+  // room server before we do any further work.
+  dispatch_semaphore_t sem = dispatch_semaphore_create(0);
+  [NSURLConnection sendAsyncRequest:request
+                  completionHandler:^(NSURLResponse *response, NSData *data, NSError *e) {
+                    if (e) {
+                      error = e;
+                    }
+                    dispatch_semaphore_signal(sem);
+                  }];
+
+  dispatch_semaphore_wait(sem, DISPATCH_TIME_FOREVER);
+  if (error) {
+    RTCLogError(@"Error leaving room %@ on room server: %@", roomId, error.localizedDescription);
+    if (completionHandler) {
+      completionHandler(error);
+    }
+    return;
+  }
+  RTCLog(@"Left room:%@ on room server.", roomId);
+  if (completionHandler) {
+    completionHandler(nil);
+  }
+}
+
+#pragma mark - Private
+
++ (NSError *)badResponseError {
+  NSError *error =
+      [[NSError alloc] initWithDomain:kARDAppEngineClientErrorDomain
+                                 code:kARDAppEngineClientErrorBadResponse
+                             userInfo:@{
+    NSLocalizedDescriptionKey: @"Error parsing response.",
+  }];
+  return error;
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDCaptureController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDCaptureController.h
new file mode 100644
index 0000000000..4febccee96
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDCaptureController.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "sdk/objc/components/capturer/RTCCameraVideoCapturer.h"
+
+@class ARDSettingsModel;
+
+// Controls the camera. Handles starting the capture, switching cameras etc.
+@interface ARDCaptureController : NSObject
+
+- (instancetype)initWithCapturer:(RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)capturer
+                        settings:(ARDSettingsModel *)settings;
+- (void)startCapture;
+- (void)startCapture:(void (^)(NSError *))completion;
+- (void)stopCapture;
+- (void)switchCamera;
+- (void)switchCamera:(void (^)(NSError *))completion;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDCaptureController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDCaptureController.m
new file mode 100644
index 0000000000..26cce9fdaa
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDCaptureController.m
@@ -0,0 +1,116 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDCaptureController.h"
+
+#import "sdk/objc/base/RTCLogging.h"
+
+#import "ARDSettingsModel.h"
+
+const Float64 kFramerateLimit = 30.0;
+
+@implementation ARDCaptureController {
+  RTC_OBJC_TYPE(RTCCameraVideoCapturer) * _capturer;
+  ARDSettingsModel *_settings;
+  BOOL _usingFrontCamera;
+}
+
+- (instancetype)initWithCapturer:(RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)capturer
+                        settings:(ARDSettingsModel *)settings {
+  if (self = [super init]) {
+    _capturer = capturer;
+    _settings = settings;
+    _usingFrontCamera = YES;
+  }
+
+  return self;
+}
+
+- (void)startCapture {
+  [self startCapture:nil];
+}
+
+- (void)startCapture:(void (^)(NSError *))completion {
+  AVCaptureDevicePosition position =
+      _usingFrontCamera ? AVCaptureDevicePositionFront : AVCaptureDevicePositionBack;
+  AVCaptureDevice *device = [self findDeviceForPosition:position];
+  AVCaptureDeviceFormat *format = [self selectFormatForDevice:device];
+
+  if (format == nil) {
+    RTCLogError(@"No valid formats for device %@", device);
+    NSAssert(NO, @"");
+
+    return;
+  }
+
+  NSInteger fps = [self selectFpsForFormat:format];
+
+  [_capturer startCaptureWithDevice:device format:format fps:fps completionHandler:completion];
+}
+
+- (void)stopCapture {
+  [_capturer stopCapture];
+}
+
+- (void)switchCamera {
+  _usingFrontCamera = !_usingFrontCamera;
+  [self startCapture:nil];
+}
+
+- (void)switchCamera:(void (^)(NSError *))completion {
+  _usingFrontCamera = !_usingFrontCamera;
+  [self startCapture:completion];
+}
+
+#pragma mark - Private
+
+- (AVCaptureDevice *)findDeviceForPosition:(AVCaptureDevicePosition)position {
+  NSArray<AVCaptureDevice *> *captureDevices =
+      [RTC_OBJC_TYPE(RTCCameraVideoCapturer) captureDevices];
+  for (AVCaptureDevice *device in captureDevices) {
+    if (device.position == position) {
+      return device;
+    }
+  }
+  return captureDevices[0];
+}
+
+- (AVCaptureDeviceFormat *)selectFormatForDevice:(AVCaptureDevice *)device {
+  NSArray<AVCaptureDeviceFormat *> *formats =
+      [RTC_OBJC_TYPE(RTCCameraVideoCapturer) supportedFormatsForDevice:device];
+  int targetWidth = [_settings currentVideoResolutionWidthFromStore];
+  int targetHeight = [_settings currentVideoResolutionHeightFromStore];
+  AVCaptureDeviceFormat *selectedFormat = nil;
+  int currentDiff = INT_MAX;
+
+  for (AVCaptureDeviceFormat *format in formats) {
+    CMVideoDimensions dimension = CMVideoFormatDescriptionGetDimensions(format.formatDescription);
+    FourCharCode pixelFormat = CMFormatDescriptionGetMediaSubType(format.formatDescription);
+    int diff = abs(targetWidth - dimension.width) + abs(targetHeight - dimension.height);
+    if (diff < currentDiff) {
+      selectedFormat = format;
+      currentDiff = diff;
+    } else if (diff == currentDiff && pixelFormat == [_capturer preferredOutputPixelFormat]) {
+      selectedFormat = format;
+    }
+  }
+
+  return selectedFormat;
+}
+
+- (NSInteger)selectFpsForFormat:(AVCaptureDeviceFormat *)format {
+  Float64 maxSupportedFramerate = 0;
+  for (AVFrameRateRange *fpsRange in format.videoSupportedFrameRateRanges) {
+    maxSupportedFramerate = fmax(maxSupportedFramerate, fpsRange.maxFrameRate);
+  }
+  return fmin(maxSupportedFramerate, kFramerateLimit);
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.h
new file mode 100644
index 0000000000..7c32c4b509
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "sdk/objc/base/RTCVideoCapturer.h"
+
+@protocol ARDExternalSampleDelegate <NSObject>
+- (void)didCaptureSampleBuffer:(CMSampleBufferRef)sampleBuffer;
+@end
+
+@interface ARDExternalSampleCapturer : RTC_OBJC_TYPE
+(RTCVideoCapturer)<ARDExternalSampleDelegate> @end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.m
new file mode 100644
index 0000000000..8bf6716ddb
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.m
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDExternalSampleCapturer.h"
+
+#import "sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.h"
+#import "sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h"
+#import "sdk/objc/base/RTCI420Buffer.h"
+#import "sdk/objc/base/RTCMutableI420Buffer.h"
+#import "sdk/objc/base/RTCMutableYUVPlanarBuffer.h"
+#import "sdk/objc/base/RTCVideoFrameBuffer.h"
+#import "sdk/objc/base/RTCYUVPlanarBuffer.h"
+#import "sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.h"
+
+@implementation ARDExternalSampleCapturer
+
+- (instancetype)initWithDelegate:(__weak id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>)delegate {
+  return [super initWithDelegate:delegate];
+}
+
+#pragma mark - ARDExternalSampleDelegate
+
+- (void)didCaptureSampleBuffer:(CMSampleBufferRef)sampleBuffer {
+  if (CMSampleBufferGetNumSamples(sampleBuffer) != 1 || !CMSampleBufferIsValid(sampleBuffer) ||
+      !CMSampleBufferDataIsReady(sampleBuffer)) {
+    return;
+  }
+
+  CVPixelBufferRef pixelBuffer = CMSampleBufferGetImageBuffer(sampleBuffer);
+  if (pixelBuffer == nil) {
+    return;
+  }
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *rtcPixelBuffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBuffer];
+  int64_t timeStampNs =
+      CMTimeGetSeconds(CMSampleBufferGetPresentationTimeStamp(sampleBuffer)) * NSEC_PER_SEC;
+  RTC_OBJC_TYPE(RTCVideoFrame) *videoFrame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:rtcPixelBuffer
+                                                  rotation:RTCVideoRotation_0
+                                               timeStampNs:timeStampNs];
+  [self.delegate capturer:self didCaptureVideoFrame:videoFrame];
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse+Internal.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse+Internal.h
new file mode 100644
index 0000000000..0edf7083c0
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse+Internal.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDJoinResponse.h"
+
+@interface ARDJoinResponse ()
+
+@property(nonatomic, assign) ARDJoinResultType result;
+@property(nonatomic, assign) BOOL isInitiator;
+@property(nonatomic, strong) NSString* roomId;
+@property(nonatomic, strong) NSString* clientId;
+@property(nonatomic, strong) NSArray* messages;
+@property(nonatomic, strong) NSURL* webSocketURL;
+@property(nonatomic, strong) NSURL* webSocketRestURL;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse.h
new file mode 100644
index 0000000000..2911202af1
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+typedef NS_ENUM(NSInteger, ARDJoinResultType) {
+  kARDJoinResultTypeUnknown,
+  kARDJoinResultTypeSuccess,
+  kARDJoinResultTypeFull
+};
+
+// Result of joining a room on the room server.
+@interface ARDJoinResponse : NSObject
+
+@property(nonatomic, readonly) ARDJoinResultType result;
+@property(nonatomic, readonly) BOOL isInitiator;
+@property(nonatomic, readonly) NSString *roomId;
+@property(nonatomic, readonly) NSString *clientId;
+@property(nonatomic, readonly) NSArray *messages;
+@property(nonatomic, readonly) NSURL *webSocketURL;
+@property(nonatomic, readonly) NSURL *webSocketRestURL;
+
++ (ARDJoinResponse *)responseFromJSONData:(NSData *)data;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse.m
new file mode 100644
index 0000000000..87d58e0db1
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDJoinResponse.m
@@ -0,0 +1,82 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDJoinResponse+Internal.h"
+
+#import "ARDSignalingMessage.h"
+#import "ARDUtilities.h"
+#import "RTCIceServer+JSON.h"
+
+static NSString const *kARDJoinResultKey = @"result";
+static NSString const *kARDJoinResultParamsKey = @"params";
+static NSString const *kARDJoinInitiatorKey = @"is_initiator";
+static NSString const *kARDJoinRoomIdKey = @"room_id";
+static NSString const *kARDJoinClientIdKey = @"client_id";
+static NSString const *kARDJoinMessagesKey = @"messages";
+static NSString const *kARDJoinWebSocketURLKey = @"wss_url";
+static NSString const *kARDJoinWebSocketRestURLKey = @"wss_post_url";
+
+@implementation ARDJoinResponse
+
+@synthesize result = _result;
+@synthesize isInitiator = _isInitiator;
+@synthesize roomId = _roomId;
+@synthesize clientId = _clientId;
+@synthesize messages = _messages;
+@synthesize webSocketURL = _webSocketURL;
+@synthesize webSocketRestURL = _webSocketRestURL;
+
++ (ARDJoinResponse *)responseFromJSONData:(NSData *)data {
+  NSDictionary *responseJSON = [NSDictionary dictionaryWithJSONData:data];
+  if (!responseJSON) {
+    return nil;
+  }
+  ARDJoinResponse *response = [[ARDJoinResponse alloc] init];
+  NSString *resultString = responseJSON[kARDJoinResultKey];
+  response.result = [[self class] resultTypeFromString:resultString];
+  NSDictionary *params = responseJSON[kARDJoinResultParamsKey];
+
+  response.isInitiator = [params[kARDJoinInitiatorKey] boolValue];
+  response.roomId = params[kARDJoinRoomIdKey];
+  response.clientId = params[kARDJoinClientIdKey];
+
+  // Parse messages.
+  NSArray *messages = params[kARDJoinMessagesKey];
+  NSMutableArray *signalingMessages =
+      [NSMutableArray arrayWithCapacity:messages.count];
+  for (NSString *message in messages) {
+    ARDSignalingMessage *signalingMessage =
+        [ARDSignalingMessage messageFromJSONString:message];
+    [signalingMessages addObject:signalingMessage];
+  }
+  response.messages = signalingMessages;
+
+  // Parse websocket urls.
+  NSString *webSocketURLString = params[kARDJoinWebSocketURLKey];
+  response.webSocketURL = [NSURL URLWithString:webSocketURLString];
+  NSString *webSocketRestURLString = params[kARDJoinWebSocketRestURLKey];
+  response.webSocketRestURL = [NSURL URLWithString:webSocketRestURLString];
+
+  return response;
+}
+
+#pragma mark - Private
+
++ (ARDJoinResultType)resultTypeFromString:(NSString *)resultString {
+  ARDJoinResultType result = kARDJoinResultTypeUnknown;
+  if ([resultString isEqualToString:@"SUCCESS"]) {
+    result = kARDJoinResultTypeSuccess;
+  } else if ([resultString isEqualToString:@"FULL"]) {
+    result = kARDJoinResultTypeFull;
+  }
+  return result;
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse+Internal.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse+Internal.h
new file mode 100644
index 0000000000..66ee76172f
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse+Internal.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDMessageResponse.h"
+
+@interface ARDMessageResponse ()
+
+@property(nonatomic, assign) ARDMessageResultType result;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse.h
new file mode 100644
index 0000000000..65468cdf78
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+typedef NS_ENUM(NSInteger, ARDMessageResultType) {
+  kARDMessageResultTypeUnknown,
+  kARDMessageResultTypeSuccess,
+  kARDMessageResultTypeInvalidRoom,
+  kARDMessageResultTypeInvalidClient
+};
+
+@interface ARDMessageResponse : NSObject
+
+@property(nonatomic, readonly) ARDMessageResultType result;
+
++ (ARDMessageResponse *)responseFromJSONData:(NSData *)data;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse.m
new file mode 100644
index 0000000000..0f5383f6d6
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDMessageResponse.m
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDMessageResponse+Internal.h"
+
+#import "ARDUtilities.h"
+
+static NSString const *kARDMessageResultKey = @"result";
+
+@implementation ARDMessageResponse
+
+@synthesize result = _result;
+
++ (ARDMessageResponse *)responseFromJSONData:(NSData *)data {
+  NSDictionary *responseJSON = [NSDictionary dictionaryWithJSONData:data];
+  if (!responseJSON) {
+    return nil;
+  }
+  ARDMessageResponse *response = [[ARDMessageResponse alloc] init];
+  response.result =
+      [[self class] resultTypeFromString:responseJSON[kARDMessageResultKey]];
+  return response;
+}
+
+#pragma mark - Private
+
++ (ARDMessageResultType)resultTypeFromString:(NSString *)resultString {
+  ARDMessageResultType result = kARDMessageResultTypeUnknown;
+  if ([resultString isEqualToString:@"SUCCESS"]) {
+    result = kARDMessageResultTypeSuccess;
+  } else if ([resultString isEqualToString:@"INVALID_CLIENT"]) {
+    result = kARDMessageResultTypeInvalidClient;
+  } else if ([resultString isEqualToString:@"INVALID_ROOM"]) {
+    result = kARDMessageResultTypeInvalidRoom;
+  }
+  return result;
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDRoomServerClient.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDRoomServerClient.h
new file mode 100644
index 0000000000..3a5818d6d6
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDRoomServerClient.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+@class ARDJoinResponse;
+@class ARDMessageResponse;
+@class ARDSignalingMessage;
+
+@protocol ARDRoomServerClient <NSObject>
+
+- (void)joinRoomWithRoomId:(NSString *)roomId
+                isLoopback:(BOOL)isLoopback
+         completionHandler:(void (^)(ARDJoinResponse *response, NSError *error))completionHandler;
+
+- (void)sendMessage:(ARDSignalingMessage *)message
+            forRoomId:(NSString *)roomId
+             clientId:(NSString *)clientId
+    completionHandler:(void (^)(ARDMessageResponse *response, NSError *error))completionHandler;
+
+- (void)leaveRoomWithRoomId:(NSString *)roomId
+                   clientId:(NSString *)clientId
+          completionHandler:(void (^)(NSError *error))completionHandler;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel+Private.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel+Private.h
new file mode 100644
index 0000000000..dc3f24ced8
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel+Private.h
@@ -0,0 +1,21 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "ARDSettingsModel.h"
+
+@class ARDSettingsStore;
+
+NS_ASSUME_NONNULL_BEGIN
+@interface ARDSettingsModel ()
+- (ARDSettingsStore *)settingsStore;
+@end
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel.h
new file mode 100644
index 0000000000..47c7defacd
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel.h
@@ -0,0 +1,123 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "sdk/objc/base/RTCVideoCodecInfo.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * Model class for user defined settings.
+ *
+ * Handles storing the settings and provides default values if setting is not
+ * set. Also provides list of available options for different settings. Stores
+ * for example video codec, video resolution and maximum bitrate.
+ */
+@interface ARDSettingsModel : NSObject
+
+/**
+ * Returns array of available capture resoultions.
+ *
+ * The capture resolutions are represented as strings in the following format
+ * [width]x[height]
+ */
+- (NSArray<NSString *> *)availableVideoResolutions;
+
+/**
+ * Returns current video resolution string.
+ * If no resolution is in store, default value of 640x480 is returned.
+ * When defaulting to value, the default is saved in store for consistency reasons.
+ */
+- (NSString *)currentVideoResolutionSettingFromStore;
+- (int)currentVideoResolutionWidthFromStore;
+- (int)currentVideoResolutionHeightFromStore;
+
+/**
+ * Stores the provided video resolution string into the store.
+ *
+ * If the provided resolution is no part of the available video resolutions
+ * the store operation will not be executed and NO will be returned.
+ * @param resolution the string to be stored.
+ * @return YES/NO depending on success.
+ */
+- (BOOL)storeVideoResolutionSetting:(NSString *)resolution;
+
+/**
+ * Returns array of available video codecs.
+ */
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)availableVideoCodecs;
+
+/**
+ * Returns current video codec setting from store if present or default (H264) otherwise.
+ */
+- (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)currentVideoCodecSettingFromStore;
+
+/**
+ * Stores the provided video codec setting into the store.
+ *
+ * If the provided video codec is not part of the available video codecs
+ * the store operation will not be executed and NO will be returned.
+ * @param video codec settings the string to be stored.
+ * @return YES/NO depending on success.
+ */
+- (BOOL)storeVideoCodecSetting:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)videoCodec;
+
+/**
+ * Returns current max bitrate setting from store if present.
+ */
+- (nullable NSNumber *)currentMaxBitrateSettingFromStore;
+
+/**
+ * Stores the provided bitrate value into the store.
+ *
+ * @param bitrate NSNumber representation of the max bitrate value.
+ */
+- (void)storeMaxBitrateSetting:(nullable NSNumber *)bitrate;
+
+/**
+ * Returns current audio only setting from store if present or default (NO) otherwise.
+ */
+- (BOOL)currentAudioOnlySettingFromStore;
+
+/**
+ * Stores the provided audio only setting into the store.
+ *
+ * @param setting the boolean value to be stored.
+ */
+- (void)storeAudioOnlySetting:(BOOL)audioOnly;
+
+/**
+ * Returns current create AecDump setting from store if present or default (NO) otherwise.
+ */
+- (BOOL)currentCreateAecDumpSettingFromStore;
+
+/**
+ * Stores the provided create AecDump setting into the store.
+ *
+ * @param setting the boolean value to be stored.
+ */
+- (void)storeCreateAecDumpSetting:(BOOL)createAecDump;
+
+/**
+ * Returns current setting whether to use manual audio config from store if present or default (YES)
+ * otherwise.
+ */
+- (BOOL)currentUseManualAudioConfigSettingFromStore;
+
+/**
+ * Stores the provided use manual audio config setting into the store.
+ *
+ * @param setting the boolean value to be stored.
+ */
+- (void)storeUseManualAudioConfigSetting:(BOOL)useManualAudioConfig;
+
+@end
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel.m
new file mode 100644
index 0000000000..9e709b0553
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsModel.m
@@ -0,0 +1,211 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDSettingsModel+Private.h"
+#import "ARDSettingsStore.h"
+
+#import "sdk/objc/api/peerconnection/RTCMediaConstraints.h"
+#import "sdk/objc/components/capturer/RTCCameraVideoCapturer.h"
+#import "sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface ARDSettingsModel () {
+  ARDSettingsStore *_settingsStore;
+}
+@end
+
+@implementation ARDSettingsModel
+
+- (NSArray<NSString *> *)availableVideoResolutions {
+  NSMutableSet<NSArray<NSNumber *> *> *resolutions =
+      [[NSMutableSet<NSArray<NSNumber *> *> alloc] init];
+  for (AVCaptureDevice *device in [RTC_OBJC_TYPE(RTCCameraVideoCapturer) captureDevices]) {
+    for (AVCaptureDeviceFormat *format in
+         [RTC_OBJC_TYPE(RTCCameraVideoCapturer) supportedFormatsForDevice:device]) {
+      CMVideoDimensions resolution =
+          CMVideoFormatDescriptionGetDimensions(format.formatDescription);
+      NSArray<NSNumber *> *resolutionObject = @[ @(resolution.width), @(resolution.height) ];
+      [resolutions addObject:resolutionObject];
+    }
+  }
+
+  NSArray<NSArray<NSNumber *> *> *sortedResolutions =
+      [[resolutions allObjects] sortedArrayUsingComparator:^NSComparisonResult(
+                                    NSArray<NSNumber *> *obj1, NSArray<NSNumber *> *obj2) {
+        NSComparisonResult cmp = [obj1.firstObject compare:obj2.firstObject];
+        if (cmp != NSOrderedSame) {
+          return cmp;
+        }
+        return [obj1.lastObject compare:obj2.lastObject];
+      }];
+
+  NSMutableArray<NSString *> *resolutionStrings = [[NSMutableArray<NSString *> alloc] init];
+  for (NSArray<NSNumber *> *resolution in sortedResolutions) {
+    NSString *resolutionString =
+        [NSString stringWithFormat:@"%@x%@", resolution.firstObject, resolution.lastObject];
+    [resolutionStrings addObject:resolutionString];
+  }
+
+  return [resolutionStrings copy];
+}
+
+- (NSString *)currentVideoResolutionSettingFromStore {
+  [self registerStoreDefaults];
+  return [[self settingsStore] videoResolution];
+}
+
+- (BOOL)storeVideoResolutionSetting:(NSString *)resolution {
+  if (![[self availableVideoResolutions] containsObject:resolution]) {
+    return NO;
+  }
+  [[self settingsStore] setVideoResolution:resolution];
+  return YES;
+}
+
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)availableVideoCodecs {
+  return [RTC_OBJC_TYPE(RTCDefaultVideoEncoderFactory) supportedCodecs];
+}
+
+- (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)currentVideoCodecSettingFromStore {
+  [self registerStoreDefaults];
+  NSData *codecData = [[self settingsStore] videoCodec];
+#if defined(WEBRTC_IOS) || __MAC_OS_X_VERSION_MIN_REQUIRED >= __MAC_10_13
+  Class expectedClass = [RTC_OBJC_TYPE(RTCVideoCodecInfo) class];
+  NSError *error;
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *videoCodecSetting =
+      [NSKeyedUnarchiver unarchivedObjectOfClass:expectedClass fromData:codecData error:&error];
+  if (!error) {
+    return videoCodecSetting;
+  }
+  return nil;
+#else
+  return [NSKeyedUnarchiver unarchiveObjectWithData:codecData];
+#endif
+}
+
+- (BOOL)storeVideoCodecSetting:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)videoCodec {
+  if (![[self availableVideoCodecs] containsObject:videoCodec]) {
+    return NO;
+  }
+
+#if defined(WEBRTC_IOS) || __MAC_OS_X_VERSION_MIN_REQUIRED >= __MAC_10_13
+  NSError *error;
+  NSData *codecData = [NSKeyedArchiver archivedDataWithRootObject:videoCodec
+                                            requiringSecureCoding:NO
+                                                            error:&error];
+  if (error) {
+    return NO;
+  }
+#else
+  NSData *codecData = [NSKeyedArchiver archivedDataWithRootObject:videoCodec];
+#endif
+
+  [[self settingsStore] setVideoCodec:codecData];
+  return YES;
+}
+
+- (nullable NSNumber *)currentMaxBitrateSettingFromStore {
+  [self registerStoreDefaults];
+  return [[self settingsStore] maxBitrate];
+}
+
+- (void)storeMaxBitrateSetting:(nullable NSNumber *)bitrate {
+  [[self settingsStore] setMaxBitrate:bitrate];
+}
+
+- (BOOL)currentAudioOnlySettingFromStore {
+  return [[self settingsStore] audioOnly];
+}
+
+- (void)storeAudioOnlySetting:(BOOL)audioOnly {
+  [[self settingsStore] setAudioOnly:audioOnly];
+}
+
+- (BOOL)currentCreateAecDumpSettingFromStore {
+  return [[self settingsStore] createAecDump];
+}
+
+- (void)storeCreateAecDumpSetting:(BOOL)createAecDump {
+  [[self settingsStore] setCreateAecDump:createAecDump];
+}
+
+- (BOOL)currentUseManualAudioConfigSettingFromStore {
+  return [[self settingsStore] useManualAudioConfig];
+}
+
+- (void)storeUseManualAudioConfigSetting:(BOOL)useManualAudioConfig {
+  [[self settingsStore] setUseManualAudioConfig:useManualAudioConfig];
+}
+
+#pragma mark - Testable
+
+- (ARDSettingsStore *)settingsStore {
+  if (!_settingsStore) {
+    _settingsStore = [[ARDSettingsStore alloc] init];
+    [self registerStoreDefaults];
+  }
+  return _settingsStore;
+}
+
+- (int)currentVideoResolutionWidthFromStore {
+  NSString *resolution = [self currentVideoResolutionSettingFromStore];
+
+  return [self videoResolutionComponentAtIndex:0 inString:resolution];
+}
+
+- (int)currentVideoResolutionHeightFromStore {
+  NSString *resolution = [self currentVideoResolutionSettingFromStore];
+  return [self videoResolutionComponentAtIndex:1 inString:resolution];
+}
+
+#pragma mark -
+
+- (NSString *)defaultVideoResolutionSetting {
+  return [self availableVideoResolutions].firstObject;
+}
+
+- (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)defaultVideoCodecSetting {
+  return [self availableVideoCodecs].firstObject;
+}
+
+- (int)videoResolutionComponentAtIndex:(int)index inString:(NSString *)resolution {
+  if (index != 0 && index != 1) {
+    return 0;
+  }
+  NSArray<NSString *> *components = [resolution componentsSeparatedByString:@"x"];
+  if (components.count != 2) {
+    return 0;
+  }
+  return components[index].intValue;
+}
+
+- (void)registerStoreDefaults {
+#if defined(WEBRTC_IOS) || __MAC_OS_X_VERSION_MIN_REQUIRED >= __MAC_10_13
+  NSError *error;
+  NSData *codecData = [NSKeyedArchiver archivedDataWithRootObject:[self defaultVideoCodecSetting]
+                                            requiringSecureCoding:NO
+                                                            error:&error];
+  if (error) {
+    return;
+  }
+#else
+  NSData *codecData = [NSKeyedArchiver archivedDataWithRootObject:[self defaultVideoCodecSetting]];
+#endif
+
+  [ARDSettingsStore setDefaultsForVideoResolution:[self defaultVideoResolutionSetting]
+                                       videoCodec:codecData
+                                          bitrate:nil
+                                        audioOnly:NO
+                                    createAecDump:NO
+                             useManualAudioConfig:YES];
+}
+@end
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsStore.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsStore.h
new file mode 100644
index 0000000000..bb051dbb26
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsStore.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * Light-weight persistent store for user settings.
+ *
+ * It will persist between application launches and application updates.
+ */
+@interface ARDSettingsStore : NSObject
+
+/**
+ * Set fallback values in case the setting has not been written by the user.
+ * @param dictionary of values to store
+ */
++ (void)setDefaultsForVideoResolution:(NSString *)videoResolution
+                           videoCodec:(NSData *)videoCodec
+                              bitrate:(nullable NSNumber *)bitrate
+                            audioOnly:(BOOL)audioOnly
+                        createAecDump:(BOOL)createAecDump
+                 useManualAudioConfig:(BOOL)useManualAudioConfig;
+
+@property(nonatomic) NSString *videoResolution;
+@property(nonatomic) NSData *videoCodec;
+
+/**
+ * Returns current max bitrate number stored in the store.
+ */
+- (nullable NSNumber *)maxBitrate;
+
+/**
+ * Stores the provided value as maximum bitrate setting.
+ * @param value the number to be stored
+ */
+- (void)setMaxBitrate:(nullable NSNumber *)value;
+
+@property(nonatomic) BOOL audioOnly;
+@property(nonatomic) BOOL createAecDump;
+@property(nonatomic) BOOL useManualAudioConfig;
+
+@end
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsStore.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsStore.m
new file mode 100644
index 0000000000..a3713e2f0e
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSettingsStore.m
@@ -0,0 +1,115 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDSettingsStore.h"
+
+static NSString *const kVideoResolutionKey = @"rtc_video_resolution_key";
+static NSString *const kVideoCodecKey = @"rtc_video_codec_info_key";
+static NSString *const kBitrateKey = @"rtc_max_bitrate_key";
+static NSString *const kAudioOnlyKey = @"rtc_audio_only_key";
+static NSString *const kCreateAecDumpKey = @"rtc_create_aec_dump_key";
+static NSString *const kUseManualAudioConfigKey = @"rtc_use_manual_audio_config_key";
+
+NS_ASSUME_NONNULL_BEGIN
+@interface ARDSettingsStore () {
+  NSUserDefaults *_storage;
+}
+@property(nonatomic, strong, readonly) NSUserDefaults *storage;
+@end
+
+@implementation ARDSettingsStore
+
++ (void)setDefaultsForVideoResolution:(NSString *)videoResolution
+                           videoCodec:(NSData *)videoCodec
+                              bitrate:(nullable NSNumber *)bitrate
+                            audioOnly:(BOOL)audioOnly
+                        createAecDump:(BOOL)createAecDump
+                 useManualAudioConfig:(BOOL)useManualAudioConfig {
+  NSMutableDictionary<NSString *, id> *defaultsDictionary = [@{
+    kAudioOnlyKey : @(audioOnly),
+    kCreateAecDumpKey : @(createAecDump),
+    kUseManualAudioConfigKey : @(useManualAudioConfig)
+  } mutableCopy];
+
+  if (videoResolution) {
+    defaultsDictionary[kVideoResolutionKey] = videoResolution;
+  }
+  if (videoCodec) {
+    defaultsDictionary[kVideoCodecKey] = videoCodec;
+  }
+  if (bitrate) {
+    defaultsDictionary[kBitrateKey] = bitrate;
+  }
+  [[NSUserDefaults standardUserDefaults] registerDefaults:defaultsDictionary];
+}
+
+- (NSUserDefaults *)storage {
+  if (!_storage) {
+    _storage = [NSUserDefaults standardUserDefaults];
+  }
+  return _storage;
+}
+
+- (NSString *)videoResolution {
+  return [self.storage objectForKey:kVideoResolutionKey];
+}
+
+- (void)setVideoResolution:(NSString *)resolution {
+  [self.storage setObject:resolution forKey:kVideoResolutionKey];
+  [self.storage synchronize];
+}
+
+- (NSData *)videoCodec {
+  return [self.storage objectForKey:kVideoCodecKey];
+}
+
+- (void)setVideoCodec:(NSData *)videoCodec {
+  [self.storage setObject:videoCodec forKey:kVideoCodecKey];
+  [self.storage synchronize];
+}
+
+- (nullable NSNumber *)maxBitrate {
+  return [self.storage objectForKey:kBitrateKey];
+}
+
+- (void)setMaxBitrate:(nullable NSNumber *)value {
+  [self.storage setObject:value forKey:kBitrateKey];
+  [self.storage synchronize];
+}
+
+- (BOOL)audioOnly {
+  return [self.storage boolForKey:kAudioOnlyKey];
+}
+
+- (void)setAudioOnly:(BOOL)audioOnly {
+  [self.storage setBool:audioOnly forKey:kAudioOnlyKey];
+  [self.storage synchronize];
+}
+
+- (BOOL)createAecDump {
+  return [self.storage boolForKey:kCreateAecDumpKey];
+}
+
+- (void)setCreateAecDump:(BOOL)createAecDump {
+  [self.storage setBool:createAecDump forKey:kCreateAecDumpKey];
+  [self.storage synchronize];
+}
+
+- (BOOL)useManualAudioConfig {
+  return [self.storage boolForKey:kUseManualAudioConfigKey];
+}
+
+- (void)setUseManualAudioConfig:(BOOL)useManualAudioConfig {
+  [self.storage setBool:useManualAudioConfig forKey:kUseManualAudioConfigKey];
+  [self.storage synchronize];
+}
+
+@end
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingChannel.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingChannel.h
new file mode 100644
index 0000000000..396b117b17
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingChannel.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "ARDSignalingMessage.h"
+
+typedef NS_ENUM(NSInteger, ARDSignalingChannelState) {
+  // State when disconnected.
+  kARDSignalingChannelStateClosed,
+  // State when connection is established but not ready for use.
+  kARDSignalingChannelStateOpen,
+  // State when connection is established and registered.
+  kARDSignalingChannelStateRegistered,
+  // State when connection encounters a fatal error.
+  kARDSignalingChannelStateError
+};
+
+@protocol ARDSignalingChannel;
+@protocol ARDSignalingChannelDelegate <NSObject>
+
+- (void)channel:(id<ARDSignalingChannel>)channel didChangeState:(ARDSignalingChannelState)state;
+
+- (void)channel:(id<ARDSignalingChannel>)channel didReceiveMessage:(ARDSignalingMessage *)message;
+
+@end
+
+@protocol ARDSignalingChannel <NSObject>
+
+@property(nonatomic, readonly) NSString *roomId;
+@property(nonatomic, readonly) NSString *clientId;
+@property(nonatomic, readonly) ARDSignalingChannelState state;
+@property(nonatomic, weak) id<ARDSignalingChannelDelegate> delegate;
+
+// Registers the channel for the given room and client id.
+- (void)registerForRoomId:(NSString *)roomId clientId:(NSString *)clientId;
+
+// Sends signaling message over the channel.
+- (void)sendMessage:(ARDSignalingMessage *)message;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.h
new file mode 100644
index 0000000000..ac19e8fba7
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "sdk/objc/api/peerconnection/RTCIceCandidate.h"
+#import "sdk/objc/api/peerconnection/RTCSessionDescription.h"
+
+typedef enum {
+  kARDSignalingMessageTypeCandidate,
+  kARDSignalingMessageTypeCandidateRemoval,
+  kARDSignalingMessageTypeOffer,
+  kARDSignalingMessageTypeAnswer,
+  kARDSignalingMessageTypeBye,
+} ARDSignalingMessageType;
+
+@interface ARDSignalingMessage : NSObject
+
+@property(nonatomic, readonly) ARDSignalingMessageType type;
+
++ (ARDSignalingMessage *)messageFromJSONString:(NSString *)jsonString;
+- (NSData *)JSONData;
+
+@end
+
+@interface ARDICECandidateMessage : ARDSignalingMessage
+
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCIceCandidate) * candidate;
+
+- (instancetype)initWithCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate;
+
+@end
+
+@interface ARDICECandidateRemovalMessage : ARDSignalingMessage
+
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *candidates;
+
+- (instancetype)initWithRemovedCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
+
+@end
+
+@interface ARDSessionDescriptionMessage : ARDSignalingMessage
+
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCSessionDescription) * sessionDescription;
+
+- (instancetype)initWithDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)description;
+
+@end
+
+@interface ARDByeMessage : ARDSignalingMessage
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.m
new file mode 100644
index 0000000000..049c0f5b0a
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.m
@@ -0,0 +1,160 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDSignalingMessage.h"
+
+#import "sdk/objc/base/RTCLogging.h"
+
+#import "ARDUtilities.h"
+#import "RTCIceCandidate+JSON.h"
+#import "RTCSessionDescription+JSON.h"
+
+static NSString * const kARDSignalingMessageTypeKey = @"type";
+static NSString * const kARDTypeValueRemoveCandidates = @"remove-candidates";
+
+@implementation ARDSignalingMessage
+
+@synthesize type = _type;
+
+- (instancetype)initWithType:(ARDSignalingMessageType)type {
+  if (self = [super init]) {
+    _type = type;
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [[NSString alloc] initWithData:[self JSONData]
+                               encoding:NSUTF8StringEncoding];
+}
+
++ (ARDSignalingMessage *)messageFromJSONString:(NSString *)jsonString {
+  NSDictionary *values = [NSDictionary dictionaryWithJSONString:jsonString];
+  if (!values) {
+    RTCLogError(@"Error parsing signaling message JSON.");
+    return nil;
+  }
+
+  NSString *typeString = values[kARDSignalingMessageTypeKey];
+  ARDSignalingMessage *message = nil;
+  if ([typeString isEqualToString:@"candidate"]) {
+    RTC_OBJC_TYPE(RTCIceCandidate) *candidate =
+        [RTC_OBJC_TYPE(RTCIceCandidate) candidateFromJSONDictionary:values];
+    message = [[ARDICECandidateMessage alloc] initWithCandidate:candidate];
+  } else if ([typeString isEqualToString:kARDTypeValueRemoveCandidates]) {
+    RTCLogInfo(@"Received remove-candidates message");
+    NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *candidates =
+        [RTC_OBJC_TYPE(RTCIceCandidate) candidatesFromJSONDictionary:values];
+    message = [[ARDICECandidateRemovalMessage alloc]
+                  initWithRemovedCandidates:candidates];
+  } else if ([typeString isEqualToString:@"offer"] ||
+             [typeString isEqualToString:@"answer"]) {
+    RTC_OBJC_TYPE(RTCSessionDescription) *description =
+        [RTC_OBJC_TYPE(RTCSessionDescription) descriptionFromJSONDictionary:values];
+    message =
+        [[ARDSessionDescriptionMessage alloc] initWithDescription:description];
+  } else if ([typeString isEqualToString:@"bye"]) {
+    message = [[ARDByeMessage alloc] init];
+  } else {
+    RTCLogError(@"Unexpected type: %@", typeString);
+  }
+  return message;
+}
+
+- (NSData *)JSONData {
+  return nil;
+}
+
+@end
+
+@implementation ARDICECandidateMessage
+
+@synthesize candidate = _candidate;
+
+- (instancetype)initWithCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate {
+  if (self = [super initWithType:kARDSignalingMessageTypeCandidate]) {
+    _candidate = candidate;
+  }
+  return self;
+}
+
+- (NSData *)JSONData {
+  return [_candidate JSONData];
+}
+
+@end
+
+@implementation ARDICECandidateRemovalMessage
+
+@synthesize candidates = _candidates;
+
+- (instancetype)initWithRemovedCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates {
+  NSParameterAssert(candidates.count);
+  if (self = [super initWithType:kARDSignalingMessageTypeCandidateRemoval]) {
+    _candidates = candidates;
+  }
+  return self;
+}
+
+- (NSData *)JSONData {
+  return [RTC_OBJC_TYPE(RTCIceCandidate) JSONDataForIceCandidates:_candidates
+                                                         withType:kARDTypeValueRemoveCandidates];
+}
+
+@end
+
+@implementation ARDSessionDescriptionMessage
+
+@synthesize sessionDescription = _sessionDescription;
+
+- (instancetype)initWithDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)description {
+  ARDSignalingMessageType messageType = kARDSignalingMessageTypeOffer;
+  RTCSdpType sdpType = description.type;
+  switch (sdpType) {
+    case RTCSdpTypeOffer:
+      messageType = kARDSignalingMessageTypeOffer;
+      break;
+    case RTCSdpTypeAnswer:
+      messageType = kARDSignalingMessageTypeAnswer;
+      break;
+    case RTCSdpTypePrAnswer:
+    case RTCSdpTypeRollback:
+      NSAssert(
+          NO, @"Unexpected type: %@", [RTC_OBJC_TYPE(RTCSessionDescription) stringForType:sdpType]);
+      break;
+  }
+  if (self = [super initWithType:messageType]) {
+    _sessionDescription = description;
+  }
+  return self;
+}
+
+- (NSData *)JSONData {
+  return [_sessionDescription JSONData];
+}
+
+@end
+
+@implementation ARDByeMessage
+
+- (instancetype)init {
+  return [super initWithType:kARDSignalingMessageTypeBye];
+}
+
+- (NSData *)JSONData {
+  NSDictionary *message = @{
+    @"type": @"bye"
+  };
+  return [NSJSONSerialization dataWithJSONObject:message
+                                         options:NSJSONWritingPrettyPrinted
+                                           error:NULL];
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.h
new file mode 100644
index 0000000000..eaffa67049
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "sdk/objc/api/peerconnection/RTCStatisticsReport.h"
+#import "sdk/objc/base/RTCMacros.h"
+
+/** Class used to accumulate stats information into a single displayable string.
+ */
+@interface ARDStatsBuilder : NSObject
+
+/** String that represents the accumulated stats reports passed into this
+ *  class.
+ */
+@property(nonatomic, readonly) NSString *statsString;
+@property(nonatomic) RTC_OBJC_TYPE(RTCStatisticsReport) * stats;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.m
new file mode 100644
index 0000000000..7ebf9fb1c7
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.m
@@ -0,0 +1,36 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDStatsBuilder.h"
+
+#import "sdk/objc/api/peerconnection/RTCLegacyStatsReport.h"
+#import "sdk/objc/base/RTCMacros.h"
+
+#import "ARDUtilities.h"
+
+@implementation ARDStatsBuilder
+
+@synthesize stats = _stats;
+
+- (NSString *)statsString {
+  NSMutableString *result = [NSMutableString string];
+
+  [result appendFormat:@"(cpu)%ld%%\n", (long)ARDGetCpuUsagePercentage()];
+
+  for (NSString *key in _stats.statistics) {
+    RTC_OBJC_TYPE(RTCStatistics) *stat = _stats.statistics[key];
+    [result appendFormat:@"%@\n", stat.description];
+  }
+
+  return result;
+}
+
+@end
+
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient+Internal.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient+Internal.h
new file mode 100644
index 0000000000..3a579f8f7f
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient+Internal.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDTURNClient.h"
+
+@interface ARDTURNClient : NSObject <ARDTURNClient>
+
+- (instancetype)initWithURL:(NSURL *)url;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient.h
new file mode 100644
index 0000000000..0399736f03
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "sdk/objc/base/RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCIceServer);
+
+@protocol ARDTURNClient <NSObject>
+
+// Returns TURN server urls if successful.
+- (void)requestServersWithCompletionHandler:(void (^)(NSArray *turnServers,
+                                                      NSError *error))completionHandler;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient.m
new file mode 100644
index 0000000000..069231cd7e
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDTURNClient.m
@@ -0,0 +1,86 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDTURNClient+Internal.h"
+
+#import "ARDUtilities.h"
+#import "RTCIceServer+JSON.h"
+
+// TODO(tkchin): move this to a configuration object.
+static NSString *kTURNRefererURLString = @"https://appr.tc";
+static NSString *kARDTURNClientErrorDomain = @"ARDTURNClient";
+static NSInteger kARDTURNClientErrorBadResponse = -1;
+
+@implementation ARDTURNClient {
+  NSURL *_url;
+}
+
+- (instancetype)initWithURL:(NSURL *)url {
+  NSParameterAssert([url absoluteString].length);
+  if (self = [super init]) {
+    _url = url;
+  }
+  return self;
+}
+
+- (void)requestServersWithCompletionHandler:
+    (void (^)(NSArray *turnServers, NSError *error))completionHandler {
+
+  NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:_url];
+  [NSURLConnection sendAsyncRequest:request
+                  completionHandler:^(NSURLResponse *response, NSData *data, NSError *error) {
+      if (error) {
+        completionHandler(nil, error);
+        return;
+      }
+      NSDictionary *responseDict = [NSDictionary dictionaryWithJSONData:data];
+      NSString *iceServerUrl = responseDict[@"ice_server_url"];
+      [self makeTurnServerRequestToURL:[NSURL URLWithString:iceServerUrl]
+                 WithCompletionHandler:completionHandler];
+    }];
+}
+
+#pragma mark - Private
+
+- (void)makeTurnServerRequestToURL:(NSURL *)url
+             WithCompletionHandler:(void (^)(NSArray *turnServers,
+                                             NSError *error))completionHandler {
+  NSMutableURLRequest *iceServerRequest = [NSMutableURLRequest requestWithURL:url];
+  iceServerRequest.HTTPMethod = @"POST";
+  [iceServerRequest addValue:kTURNRefererURLString forHTTPHeaderField:@"referer"];
+  [NSURLConnection sendAsyncRequest:iceServerRequest
+                  completionHandler:^(NSURLResponse *response,
+                                      NSData *data,
+                                      NSError *error) {
+      if (error) {
+        completionHandler(nil, error);
+        return;
+      }
+      NSDictionary *turnResponseDict = [NSDictionary dictionaryWithJSONData:data];
+      NSMutableArray *turnServers = [NSMutableArray array];
+      [turnResponseDict[@"iceServers"]
+          enumerateObjectsUsingBlock:^(NSDictionary *obj, NSUInteger idx, BOOL *stop) {
+            [turnServers addObject:[RTC_OBJC_TYPE(RTCIceServer) serverFromJSONDictionary:obj]];
+          }];
+      if (!turnServers) {
+        NSError *responseError =
+          [[NSError alloc] initWithDomain:kARDTURNClientErrorDomain
+                                     code:kARDTURNClientErrorBadResponse
+                                 userInfo:@{
+            NSLocalizedDescriptionKey: @"Bad TURN response.",
+            }];
+        completionHandler(nil, responseError);
+        return;
+      }
+      completionHandler(turnServers, nil);
+    }];
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.h
new file mode 100644
index 0000000000..81888e6e83
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "ARDSignalingChannel.h"
+
+// Wraps a WebSocket connection to the AppRTC WebSocket server.
+@interface ARDWebSocketChannel : NSObject <ARDSignalingChannel>
+
+- (instancetype)initWithURL:(NSURL *)url
+                    restURL:(NSURL *)restURL
+                   delegate:(id<ARDSignalingChannelDelegate>)delegate;
+
+// Registers with the WebSocket server for the given room and client id once
+// the web socket connection is open.
+- (void)registerForRoomId:(NSString *)roomId clientId:(NSString *)clientId;
+
+// Sends message over the WebSocket connection if registered, otherwise POSTs to
+// the web socket server instead.
+- (void)sendMessage:(ARDSignalingMessage *)message;
+
+@end
+
+// Loopback mode is used to cause the client to connect to itself for testing.
+// A second web socket connection is established simulating the other client.
+// Any messages received are sent back to the WebSocket server after modifying
+// them as appropriate.
+@interface ARDLoopbackWebSocketChannel : ARDWebSocketChannel
+
+- (instancetype)initWithURL:(NSURL *)url restURL:(NSURL *)restURL;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.m
new file mode 100644
index 0000000000..bbb0bf87f8
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.m
@@ -0,0 +1,252 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDWebSocketChannel.h"
+
+#import "sdk/objc/base/RTCLogging.h"
+
+#import "SRWebSocket.h"
+
+#import "ARDSignalingMessage.h"
+#import "ARDUtilities.h"
+
+// TODO(tkchin): move these to a configuration object.
+static NSString const *kARDWSSMessageErrorKey = @"error";
+static NSString const *kARDWSSMessagePayloadKey = @"msg";
+
+@interface ARDWebSocketChannel () <SRWebSocketDelegate>
+@end
+
+@implementation ARDWebSocketChannel {
+  NSURL *_url;
+  NSURL *_restURL;
+  SRWebSocket *_socket;
+}
+
+@synthesize delegate = _delegate;
+@synthesize state = _state;
+@synthesize roomId = _roomId;
+@synthesize clientId = _clientId;
+
+- (instancetype)initWithURL:(NSURL *)url
+                    restURL:(NSURL *)restURL
+                   delegate:(id<ARDSignalingChannelDelegate>)delegate {
+  if (self = [super init]) {
+    _url = url;
+    _restURL = restURL;
+    _delegate = delegate;
+    _socket = [[SRWebSocket alloc] initWithURL:url];
+    _socket.delegate = self;
+    RTCLog(@"Opening WebSocket.");
+    [_socket open];
+  }
+  return self;
+}
+
+- (void)dealloc {
+  [self disconnect];
+}
+
+- (void)setState:(ARDSignalingChannelState)state {
+  if (_state == state) {
+    return;
+  }
+  _state = state;
+  [_delegate channel:self didChangeState:_state];
+}
+
+- (void)registerForRoomId:(NSString *)roomId
+                 clientId:(NSString *)clientId {
+  NSParameterAssert(roomId.length);
+  NSParameterAssert(clientId.length);
+  _roomId = roomId;
+  _clientId = clientId;
+  if (_state == kARDSignalingChannelStateOpen) {
+    [self registerWithCollider];
+  }
+}
+
+- (void)sendMessage:(ARDSignalingMessage *)message {
+  NSParameterAssert(_clientId.length);
+  NSParameterAssert(_roomId.length);
+  NSData *data = [message JSONData];
+  if (_state == kARDSignalingChannelStateRegistered) {
+    NSString *payload =
+        [[NSString alloc] initWithData:data encoding:NSUTF8StringEncoding];
+    NSDictionary *message = @{
+      @"cmd": @"send",
+      @"msg": payload,
+    };
+    NSData *messageJSONObject =
+        [NSJSONSerialization dataWithJSONObject:message
+                                        options:NSJSONWritingPrettyPrinted
+                                          error:nil];
+    NSString *messageString =
+        [[NSString alloc] initWithData:messageJSONObject
+                              encoding:NSUTF8StringEncoding];
+    RTCLog(@"C->WSS: %@", messageString);
+    [_socket send:messageString];
+  } else {
+    NSString *dataString =
+        [[NSString alloc] initWithData:data encoding:NSUTF8StringEncoding];
+    RTCLog(@"C->WSS POST: %@", dataString);
+    NSString *urlString =
+        [NSString stringWithFormat:@"%@/%@/%@",
+            [_restURL absoluteString], _roomId, _clientId];
+    NSURL *url = [NSURL URLWithString:urlString];
+    [NSURLConnection sendAsyncPostToURL:url
+                               withData:data
+                      completionHandler:nil];
+  }
+}
+
+- (void)disconnect {
+  if (_state == kARDSignalingChannelStateClosed ||
+      _state == kARDSignalingChannelStateError) {
+    return;
+  }
+  [_socket close];
+  RTCLog(@"C->WSS DELETE rid:%@ cid:%@", _roomId, _clientId);
+  NSString *urlString =
+      [NSString stringWithFormat:@"%@/%@/%@",
+          [_restURL absoluteString], _roomId, _clientId];
+  NSURL *url = [NSURL URLWithString:urlString];
+  NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:url];
+  request.HTTPMethod = @"DELETE";
+  request.HTTPBody = nil;
+  [NSURLConnection sendAsyncRequest:request completionHandler:nil];
+}
+
+#pragma mark - SRWebSocketDelegate
+
+- (void)webSocketDidOpen:(SRWebSocket *)webSocket {
+  RTCLog(@"WebSocket connection opened.");
+  self.state = kARDSignalingChannelStateOpen;
+  if (_roomId.length && _clientId.length) {
+    [self registerWithCollider];
+  }
+}
+
+- (void)webSocket:(SRWebSocket *)webSocket didReceiveMessage:(id)message {
+  NSString *messageString = message;
+  NSData *messageData = [messageString dataUsingEncoding:NSUTF8StringEncoding];
+  id jsonObject = [NSJSONSerialization JSONObjectWithData:messageData
+                                                  options:0
+                                                    error:nil];
+  if (![jsonObject isKindOfClass:[NSDictionary class]]) {
+    RTCLogError(@"Unexpected message: %@", jsonObject);
+    return;
+  }
+  NSDictionary *wssMessage = jsonObject;
+  NSString *errorString = wssMessage[kARDWSSMessageErrorKey];
+  if (errorString.length) {
+    RTCLogError(@"WSS error: %@", errorString);
+    return;
+  }
+  NSString *payload = wssMessage[kARDWSSMessagePayloadKey];
+  ARDSignalingMessage *signalingMessage =
+      [ARDSignalingMessage messageFromJSONString:payload];
+  RTCLog(@"WSS->C: %@", payload);
+  [_delegate channel:self didReceiveMessage:signalingMessage];
+}
+
+- (void)webSocket:(SRWebSocket *)webSocket didFailWithError:(NSError *)error {
+  RTCLogError(@"WebSocket error: %@", error);
+  self.state = kARDSignalingChannelStateError;
+}
+
+- (void)webSocket:(SRWebSocket *)webSocket
+    didCloseWithCode:(NSInteger)code
+              reason:(NSString *)reason
+            wasClean:(BOOL)wasClean {
+  RTCLog(@"WebSocket closed with code: %ld reason:%@ wasClean:%d",
+      (long)code, reason, wasClean);
+  NSParameterAssert(_state != kARDSignalingChannelStateError);
+  self.state = kARDSignalingChannelStateClosed;
+}
+
+#pragma mark - Private
+
+- (void)registerWithCollider {
+  if (_state == kARDSignalingChannelStateRegistered) {
+    return;
+  }
+  NSParameterAssert(_roomId.length);
+  NSParameterAssert(_clientId.length);
+  NSDictionary *registerMessage = @{
+    @"cmd": @"register",
+    @"roomid" : _roomId,
+    @"clientid" : _clientId,
+  };
+  NSData *message =
+      [NSJSONSerialization dataWithJSONObject:registerMessage
+                                      options:NSJSONWritingPrettyPrinted
+                                        error:nil];
+  NSString *messageString =
+      [[NSString alloc] initWithData:message encoding:NSUTF8StringEncoding];
+  RTCLog(@"Registering on WSS for rid:%@ cid:%@", _roomId, _clientId);
+  // Registration can fail if server rejects it. For example, if the room is
+  // full.
+  [_socket send:messageString];
+  self.state = kARDSignalingChannelStateRegistered;
+}
+
+@end
+
+@interface ARDLoopbackWebSocketChannel () <ARDSignalingChannelDelegate>
+@end
+
+@implementation ARDLoopbackWebSocketChannel
+
+- (instancetype)initWithURL:(NSURL *)url restURL:(NSURL *)restURL {
+  return [super initWithURL:url restURL:restURL delegate:self];
+}
+
+#pragma mark - ARDSignalingChannelDelegate
+
+- (void)channel:(id<ARDSignalingChannel>)channel
+    didReceiveMessage:(ARDSignalingMessage *)message {
+  switch (message.type) {
+    case kARDSignalingMessageTypeOffer: {
+      // Change message to answer, send back to server.
+      ARDSessionDescriptionMessage *sdpMessage =
+          (ARDSessionDescriptionMessage *)message;
+      RTC_OBJC_TYPE(RTCSessionDescription) *description = sdpMessage.sessionDescription;
+      NSString *dsc = description.sdp;
+      dsc = [dsc stringByReplacingOccurrencesOfString:@"offer"
+                                           withString:@"answer"];
+      RTC_OBJC_TYPE(RTCSessionDescription) *answerDescription =
+          [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithType:RTCSdpTypeAnswer sdp:dsc];
+      ARDSignalingMessage *answer =
+          [[ARDSessionDescriptionMessage alloc]
+               initWithDescription:answerDescription];
+      [self sendMessage:answer];
+      break;
+    }
+    case kARDSignalingMessageTypeAnswer:
+      // Should not receive answer in loopback scenario.
+      break;
+    case kARDSignalingMessageTypeCandidate:
+    case kARDSignalingMessageTypeCandidateRemoval:
+      // Send back to server.
+      [self sendMessage:message];
+      break;
+    case kARDSignalingMessageTypeBye:
+      // Nothing to do.
+      return;
+  }
+}
+
+- (void)channel:(id<ARDSignalingChannel>)channel
+    didChangeState:(ARDSignalingChannelState)state {
+}
+
+@end
+
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.h
new file mode 100644
index 0000000000..5fd823f2de
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "sdk/objc/api/peerconnection/RTCIceCandidate.h"
+
+@interface RTC_OBJC_TYPE (RTCIceCandidate)
+(JSON)
+
+    + (RTC_OBJC_TYPE(RTCIceCandidate) *)candidateFromJSONDictionary : (NSDictionary *)dictionary;
++ (NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidatesFromJSONDictionary:
+    (NSDictionary *)dictionary;
++ (NSData *)JSONDataForIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates
+                            withType:(NSString *)typeValue;
+- (NSData *)JSONData;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.m
new file mode 100644
index 0000000000..99cefbff0b
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.m
@@ -0,0 +1,100 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceCandidate+JSON.h"
+
+#import "sdk/objc/base/RTCLogging.h"
+
+static NSString const *kRTCICECandidateTypeKey = @"type";
+static NSString const *kRTCICECandidateTypeValue = @"candidate";
+static NSString const *kRTCICECandidateMidKey = @"id";
+static NSString const *kRTCICECandidateMLineIndexKey = @"label";
+static NSString const *kRTCICECandidateSdpKey = @"candidate";
+static NSString const *kRTCICECandidatesTypeKey = @"candidates";
+
+@implementation RTC_OBJC_TYPE (RTCIceCandidate)
+(JSON)
+
+    + (RTC_OBJC_TYPE(RTCIceCandidate) *)candidateFromJSONDictionary : (NSDictionary *)dictionary {
+  NSString *mid = dictionary[kRTCICECandidateMidKey];
+  NSString *sdp = dictionary[kRTCICECandidateSdpKey];
+  NSNumber *num = dictionary[kRTCICECandidateMLineIndexKey];
+  NSInteger mLineIndex = [num integerValue];
+  return [[RTC_OBJC_TYPE(RTCIceCandidate) alloc] initWithSdp:sdp
+                                               sdpMLineIndex:mLineIndex
+                                                      sdpMid:mid];
+}
+
++ (NSData *)JSONDataForIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates
+                            withType:(NSString *)typeValue {
+  NSMutableArray *jsonCandidates =
+      [NSMutableArray arrayWithCapacity:candidates.count];
+  for (RTC_OBJC_TYPE(RTCIceCandidate) * candidate in candidates) {
+    NSDictionary *jsonCandidate = [candidate JSONDictionary];
+    [jsonCandidates addObject:jsonCandidate];
+  }
+  NSDictionary *json = @{
+    kRTCICECandidateTypeKey : typeValue,
+    kRTCICECandidatesTypeKey : jsonCandidates
+  };
+  NSError *error = nil;
+  NSData *data =
+      [NSJSONSerialization dataWithJSONObject:json
+                                      options:NSJSONWritingPrettyPrinted
+                                        error:&error];
+  if (error) {
+    RTCLogError(@"Error serializing JSON: %@", error);
+    return nil;
+  }
+  return data;
+}
+
++ (NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidatesFromJSONDictionary:
+    (NSDictionary *)dictionary {
+  NSArray *jsonCandidates = dictionary[kRTCICECandidatesTypeKey];
+  NSMutableArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *candidates =
+      [NSMutableArray arrayWithCapacity:jsonCandidates.count];
+  for (NSDictionary *jsonCandidate in jsonCandidates) {
+    RTC_OBJC_TYPE(RTCIceCandidate) *candidate =
+        [RTC_OBJC_TYPE(RTCIceCandidate) candidateFromJSONDictionary:jsonCandidate];
+    [candidates addObject:candidate];
+  }
+  return candidates;
+}
+
+- (NSData *)JSONData {
+  NSDictionary *json = @{
+    kRTCICECandidateTypeKey : kRTCICECandidateTypeValue,
+    kRTCICECandidateMLineIndexKey : @(self.sdpMLineIndex),
+    kRTCICECandidateMidKey : self.sdpMid,
+    kRTCICECandidateSdpKey : self.sdp
+  };
+  NSError *error = nil;
+  NSData *data =
+      [NSJSONSerialization dataWithJSONObject:json
+                                      options:NSJSONWritingPrettyPrinted
+                                        error:&error];
+  if (error) {
+    RTCLogError(@"Error serializing JSON: %@", error);
+    return nil;
+  }
+  return data;
+}
+
+- (NSDictionary *)JSONDictionary{
+  NSDictionary *json = @{
+    kRTCICECandidateMLineIndexKey : @(self.sdpMLineIndex),
+    kRTCICECandidateMidKey : self.sdpMid,
+    kRTCICECandidateSdpKey : self.sdp
+  };
+  return json;
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceServer+JSON.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceServer+JSON.h
new file mode 100644
index 0000000000..35f6af7583
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceServer+JSON.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "sdk/objc/api/peerconnection/RTCIceServer.h"
+
+@interface RTC_OBJC_TYPE (RTCIceServer)
+(JSON)
+
+    + (RTC_OBJC_TYPE(RTCIceServer) *)serverFromJSONDictionary : (NSDictionary *)dictionary;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceServer+JSON.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceServer+JSON.m
new file mode 100644
index 0000000000..b5272a2f64
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCIceServer+JSON.m
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceServer+JSON.h"
+
+@implementation RTC_OBJC_TYPE (RTCIceServer)
+(JSON)
+
+    + (RTC_OBJC_TYPE(RTCIceServer) *)serverFromJSONDictionary : (NSDictionary *)dictionary {
+  NSArray *turnUrls = dictionary[@"urls"];
+  NSString *username = dictionary[@"username"] ?: @"";
+  NSString *credential = dictionary[@"credential"] ?: @"";
+  return [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:turnUrls
+                                                        username:username
+                                                      credential:credential];
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.h
new file mode 100644
index 0000000000..74d03d11b9
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.h
@@ -0,0 +1,20 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "sdk/objc/api/peerconnection/RTCSessionDescription.h"
+
+@interface RTC_OBJC_TYPE (RTCSessionDescription)
+(JSON)
+
+    + (RTC_OBJC_TYPE(RTCSessionDescription) *)descriptionFromJSONDictionary
+    : (NSDictionary *)dictionary;
+- (NSData *)JSONData;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.m
new file mode 100644
index 0000000000..28268faa84
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.m
@@ -0,0 +1,36 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCSessionDescription+JSON.h"
+
+static NSString const *kRTCSessionDescriptionTypeKey = @"type";
+static NSString const *kRTCSessionDescriptionSdpKey = @"sdp";
+
+@implementation RTC_OBJC_TYPE (RTCSessionDescription)
+(JSON)
+
+    + (RTC_OBJC_TYPE(RTCSessionDescription) *)descriptionFromJSONDictionary
+    : (NSDictionary *)dictionary {
+  NSString *typeString = dictionary[kRTCSessionDescriptionTypeKey];
+  RTCSdpType type = [[self class] typeForString:typeString];
+  NSString *sdp = dictionary[kRTCSessionDescriptionSdpKey];
+  return [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithType:type sdp:sdp];
+}
+
+- (NSData *)JSONData {
+  NSString *type = [[self class] stringForType:self.type];
+  NSDictionary *json = @{
+    kRTCSessionDescriptionTypeKey : type,
+    kRTCSessionDescriptionSdpKey : self.sdp
+  };
+  return [NSJSONSerialization dataWithJSONObject:json options:0 error:nil];
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/common/ARDUtilities.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/common/ARDUtilities.h
new file mode 100644
index 0000000000..5f0d7dbef7
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/common/ARDUtilities.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+@interface NSDictionary (ARDUtilites)
+
+// Creates a dictionary with the keys and values in the JSON object.
++ (NSDictionary *)dictionaryWithJSONString:(NSString *)jsonString;
++ (NSDictionary *)dictionaryWithJSONData:(NSData *)jsonData;
+
+@end
+
+@interface NSURLConnection (ARDUtilities)
+
+// Issues an asynchronous request that calls back on main queue.
++ (void)sendAsyncRequest:(NSURLRequest *)request
+       completionHandler:
+           (void (^)(NSURLResponse *response, NSData *data, NSError *error))completionHandler;
+
+// Posts data to the specified URL.
++ (void)sendAsyncPostToURL:(NSURL *)url
+                  withData:(NSData *)data
+         completionHandler:(void (^)(BOOL succeeded, NSData *data))completionHandler;
+
+@end
+
+NSInteger ARDGetCpuUsagePercentage(void);
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/common/ARDUtilities.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/common/ARDUtilities.m
new file mode 100644
index 0000000000..e0674f5210
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/common/ARDUtilities.m
@@ -0,0 +1,126 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDUtilities.h"
+
+#import <mach/mach.h>
+
+#import "sdk/objc/base/RTCLogging.h"
+
+@implementation NSDictionary (ARDUtilites)
+
++ (NSDictionary *)dictionaryWithJSONString:(NSString *)jsonString {
+  NSParameterAssert(jsonString.length > 0);
+  NSData *data = [jsonString dataUsingEncoding:NSUTF8StringEncoding];
+  NSError *error = nil;
+  NSDictionary *dict =
+      [NSJSONSerialization JSONObjectWithData:data options:0 error:&error];
+  if (error) {
+    RTCLogError(@"Error parsing JSON: %@", error.localizedDescription);
+  }
+  return dict;
+}
+
++ (NSDictionary *)dictionaryWithJSONData:(NSData *)jsonData {
+  NSError *error = nil;
+  NSDictionary *dict =
+      [NSJSONSerialization JSONObjectWithData:jsonData options:0 error:&error];
+  if (error) {
+    RTCLogError(@"Error parsing JSON: %@", error.localizedDescription);
+  }
+  return dict;
+}
+
+@end
+
+@implementation NSURLConnection (ARDUtilities)
+
++ (void)sendAsyncRequest:(NSURLRequest *)request
+       completionHandler:(void (^)(NSURLResponse *response,
+                                   NSData *data,
+                                   NSError *error))completionHandler {
+  // Kick off an async request which will call back on main thread.
+  NSURLSession *session = [NSURLSession sharedSession];
+  [[session dataTaskWithRequest:request
+              completionHandler:^(NSData *data, NSURLResponse *response, NSError *error) {
+                if (completionHandler) {
+                  completionHandler(response, data, error);
+                }
+              }] resume];
+}
+
+// Posts data to the specified URL.
++ (void)sendAsyncPostToURL:(NSURL *)url
+                  withData:(NSData *)data
+         completionHandler:(void (^)(BOOL succeeded,
+                                     NSData *data))completionHandler {
+  NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:url];
+  request.HTTPMethod = @"POST";
+  request.HTTPBody = data;
+  [[self class] sendAsyncRequest:request
+                completionHandler:^(NSURLResponse *response,
+                                    NSData *data,
+                                    NSError *error) {
+    if (error) {
+      RTCLogError(@"Error posting data: %@", error.localizedDescription);
+      if (completionHandler) {
+        completionHandler(NO, data);
+      }
+      return;
+    }
+    NSHTTPURLResponse *httpResponse = (NSHTTPURLResponse *)response;
+    if (httpResponse.statusCode != 200) {
+      NSString *serverResponse = data.length > 0 ?
+          [[NSString alloc] initWithData:data encoding:NSUTF8StringEncoding] :
+          nil;
+      RTCLogError(@"Received bad response: %@", serverResponse);
+      if (completionHandler) {
+        completionHandler(NO, data);
+      }
+      return;
+    }
+    if (completionHandler) {
+      completionHandler(YES, data);
+    }
+  }];
+}
+
+@end
+
+NSInteger ARDGetCpuUsagePercentage(void) {
+  // Create an array of thread ports for the current task.
+  const task_t task = mach_task_self();
+  thread_act_array_t thread_array;
+  mach_msg_type_number_t thread_count;
+  if (task_threads(task, &thread_array, &thread_count) != KERN_SUCCESS) {
+    return -1;
+  }
+
+  // Sum cpu usage from all threads.
+  float cpu_usage_percentage = 0;
+  thread_basic_info_data_t thread_info_data = {};
+  mach_msg_type_number_t thread_info_count;
+  for (size_t i = 0; i < thread_count; ++i) {
+    thread_info_count = THREAD_BASIC_INFO_COUNT;
+    kern_return_t ret = thread_info(thread_array[i],
+                                    THREAD_BASIC_INFO,
+                                    (thread_info_t)&thread_info_data,
+                                    &thread_info_count);
+    if (ret == KERN_SUCCESS) {
+      cpu_usage_percentage +=
+          100.f * (float)thread_info_data.cpu_usage / TH_USAGE_SCALE;
+    }
+  }
+
+  // Dealloc the created array.
+  vm_deallocate(task, (vm_address_t)thread_array,
+                sizeof(thread_act_t) * thread_count);
+  return lroundf(cpu_usage_percentage);
+}
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.h
new file mode 100644
index 0000000000..7eafff8ebc
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2013 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+// The main application class of the AppRTCMobile iOS app demonstrating
+// interoperability between the Objective C implementation of PeerConnection
+// and the appr.tc demo webapp.
+@interface ARDAppDelegate : NSObject <UIApplicationDelegate>
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m
new file mode 100644
index 0000000000..51e9910b87
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m
@@ -0,0 +1,56 @@
+/*
+ *  Copyright 2013 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDAppDelegate.h"
+
+#import "sdk/objc/api/peerconnection/RTCFieldTrials.h"
+#import "sdk/objc/api/peerconnection/RTCSSLAdapter.h"
+#import "sdk/objc/api/peerconnection/RTCTracing.h"
+#import "sdk/objc/base/RTCLogging.h"
+
+#import "ARDMainViewController.h"
+
+@implementation ARDAppDelegate {
+  UIWindow *_window;
+}
+
+#pragma mark - UIApplicationDelegate methods
+
+- (BOOL)application:(UIApplication *)application
+    didFinishLaunchingWithOptions:(NSDictionary *)launchOptions {
+  NSDictionary *fieldTrials = @{};
+  RTCInitFieldTrialDictionary(fieldTrials);
+  RTCInitializeSSL();
+  RTCSetupInternalTracer();
+  _window =  [[UIWindow alloc] initWithFrame:[[UIScreen mainScreen] bounds]];
+  [_window makeKeyAndVisible];
+  ARDMainViewController *viewController = [[ARDMainViewController alloc] init];
+
+  UINavigationController *root =
+      [[UINavigationController alloc] initWithRootViewController:viewController];
+  root.navigationBar.translucent = NO;
+  _window.rootViewController = root;
+
+#if defined(NDEBUG)
+  // In debug builds the default level is LS_INFO and in non-debug builds it is
+  // disabled. Continue to log to console in non-debug builds, but only
+  // warnings and errors.
+  RTCSetMinDebugLogLevel(RTCLoggingSeverityWarning);
+#endif
+
+  return YES;
+}
+
+- (void)applicationWillTerminate:(UIApplication *)application {
+  RTCShutdownInternalTracer();
+  RTCCleanupSSL();
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDFileCaptureController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDFileCaptureController.h
new file mode 100644
index 0000000000..82f8fcdd1b
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDFileCaptureController.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "sdk/objc/base/RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCFileVideoCapturer);
+
+/**
+ * Controls a file capturer.
+ */
+NS_CLASS_AVAILABLE_IOS(10)
+@interface ARDFileCaptureController : NSObject
+
+/**
+ * Creates instance of the controller.
+ *
+ * @param capturer The capturer to be controlled.
+ */
+- (instancetype)initWithCapturer:(RTC_OBJC_TYPE(RTCFileVideoCapturer) *)capturer;
+
+/**
+ * Starts the file capturer.
+ *
+ * Possible errors produced by the capturer will be logged.
+ */
+- (void)startCapture;
+
+/**
+ * Immediately stops capturer.
+ */
+- (void)stopCapture;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDFileCaptureController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDFileCaptureController.m
new file mode 100644
index 0000000000..2ddde6dd59
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDFileCaptureController.m
@@ -0,0 +1,45 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDFileCaptureController.h"
+
+#import "sdk/objc/components/capturer/RTCFileVideoCapturer.h"
+
+@interface ARDFileCaptureController ()
+
+@property(nonatomic, strong) RTC_OBJC_TYPE(RTCFileVideoCapturer) * fileCapturer;
+
+@end
+
+@implementation ARDFileCaptureController
+@synthesize fileCapturer = _fileCapturer;
+
+- (instancetype)initWithCapturer:(RTC_OBJC_TYPE(RTCFileVideoCapturer) *)capturer {
+  if (self = [super init]) {
+    _fileCapturer = capturer;
+  }
+  return self;
+}
+
+- (void)startCapture {
+  [self startFileCapture];
+}
+
+- (void)startFileCapture {
+  [self.fileCapturer startCapturingFromFileNamed:@"foreman.mp4"
+                                         onError:^(NSError *_Nonnull error) {
+                                           NSLog(@"Error %@", error.userInfo);
+                                         }];
+}
+
+- (void)stopCapture {
+  [self.fileCapturer stopCapture];
+}
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainView.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainView.h
new file mode 100644
index 0000000000..c6691c2d84
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainView.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+@class ARDMainView;
+
+@protocol ARDMainViewDelegate <NSObject>
+
+- (void)mainView:(ARDMainView *)mainView didInputRoom:(NSString *)room isLoopback:(BOOL)isLoopback;
+- (void)mainViewDidToggleAudioLoop:(ARDMainView *)mainView;
+
+@end
+
+// The main view of AppRTCMobile. It contains an input field for entering a room
+// name on apprtc to connect to.
+@interface ARDMainView : UIView
+
+@property(nonatomic, weak) id<ARDMainViewDelegate> delegate;
+// Updates the audio loop button as needed.
+@property(nonatomic, assign) BOOL isAudioLoopPlaying;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainView.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainView.m
new file mode 100644
index 0000000000..d9521060eb
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainView.m
@@ -0,0 +1,196 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDMainView.h"
+
+#import "UIImage+ARDUtilities.h"
+
+static CGFloat const kRoomTextFieldHeight = 40;
+static CGFloat const kRoomTextFieldMargin = 8;
+static CGFloat const kCallControlMargin = 8;
+
+// Helper view that contains a text field and a clear button.
+@interface ARDRoomTextField : UIView <UITextFieldDelegate>
+@property(nonatomic, readonly) NSString *roomText;
+@end
+
+@implementation ARDRoomTextField {
+  UITextField *_roomText;
+}
+
+- (instancetype)initWithFrame:(CGRect)frame {
+  if (self = [super initWithFrame:frame]) {
+    _roomText = [[UITextField alloc] initWithFrame:CGRectZero];
+    _roomText.borderStyle = UITextBorderStyleNone;
+    _roomText.font = [UIFont systemFontOfSize:12];
+    _roomText.placeholder = @"Room name";
+    _roomText.autocorrectionType = UITextAutocorrectionTypeNo;
+    _roomText.autocapitalizationType = UITextAutocapitalizationTypeNone;
+    _roomText.clearButtonMode = UITextFieldViewModeAlways;
+    _roomText.delegate = self;
+    [self addSubview:_roomText];
+
+    // Give rounded corners and a light gray border.
+    self.layer.borderWidth = 1;
+    self.layer.borderColor = [[UIColor lightGrayColor] CGColor];
+    self.layer.cornerRadius = 2;
+  }
+  return self;
+}
+
+- (void)layoutSubviews {
+  _roomText.frame =
+      CGRectMake(kRoomTextFieldMargin, 0, CGRectGetWidth(self.bounds) - kRoomTextFieldMargin,
+                 kRoomTextFieldHeight);
+}
+
+- (CGSize)sizeThatFits:(CGSize)size {
+  size.height = kRoomTextFieldHeight;
+  return size;
+}
+
+- (NSString *)roomText {
+  return _roomText.text;
+}
+
+#pragma mark - UITextFieldDelegate
+
+- (BOOL)textFieldShouldReturn:(UITextField *)textField {
+  // There is no other control that can take focus, so manually resign focus
+  // when return (Join) is pressed to trigger `textFieldDidEndEditing`.
+  [textField resignFirstResponder];
+  return YES;
+}
+
+@end
+
+@implementation ARDMainView {
+  ARDRoomTextField *_roomText;
+  UIButton *_startRegularCallButton;
+  UIButton *_startLoopbackCallButton;
+  UIButton *_audioLoopButton;
+}
+
+@synthesize delegate = _delegate;
+@synthesize isAudioLoopPlaying = _isAudioLoopPlaying;
+
+- (instancetype)initWithFrame:(CGRect)frame {
+  if (self = [super initWithFrame:frame]) {
+    _roomText = [[ARDRoomTextField alloc] initWithFrame:CGRectZero];
+    [self addSubview:_roomText];
+
+    UIFont *controlFont = [UIFont boldSystemFontOfSize:18.0];
+    UIColor *controlFontColor = [UIColor whiteColor];
+
+    _startRegularCallButton = [UIButton buttonWithType:UIButtonTypeSystem];
+    _startRegularCallButton.titleLabel.font = controlFont;
+    [_startRegularCallButton setTitleColor:controlFontColor forState:UIControlStateNormal];
+    _startRegularCallButton.backgroundColor
+        = [UIColor colorWithRed:66.0/255.0 green:200.0/255.0 blue:90.0/255.0 alpha:1.0];
+    [_startRegularCallButton setTitle:@"Call room" forState:UIControlStateNormal];
+    [_startRegularCallButton addTarget:self
+                                action:@selector(onStartRegularCall:)
+                      forControlEvents:UIControlEventTouchUpInside];
+    [self addSubview:_startRegularCallButton];
+
+    _startLoopbackCallButton = [UIButton buttonWithType:UIButtonTypeSystem];
+    _startLoopbackCallButton.titleLabel.font = controlFont;
+    [_startLoopbackCallButton setTitleColor:controlFontColor forState:UIControlStateNormal];
+    _startLoopbackCallButton.backgroundColor =
+        [UIColor colorWithRed:0.0 green:122.0/255.0 blue:1.0 alpha:1.0];
+    [_startLoopbackCallButton setTitle:@"Loopback call" forState:UIControlStateNormal];
+    [_startLoopbackCallButton addTarget:self
+                                 action:@selector(onStartLoopbackCall:)
+                       forControlEvents:UIControlEventTouchUpInside];
+    [self addSubview:_startLoopbackCallButton];
+
+
+    // Used to test what happens to sounds when calls are in progress.
+    _audioLoopButton = [UIButton buttonWithType:UIButtonTypeSystem];
+    _audioLoopButton.titleLabel.font = controlFont;
+    [_audioLoopButton setTitleColor:controlFontColor forState:UIControlStateNormal];
+    _audioLoopButton.backgroundColor =
+        [UIColor colorWithRed:1.0 green:149.0/255.0 blue:0.0 alpha:1.0];
+    [self updateAudioLoopButton];
+    [_audioLoopButton addTarget:self
+                         action:@selector(onToggleAudioLoop:)
+               forControlEvents:UIControlEventTouchUpInside];
+    [self addSubview:_audioLoopButton];
+
+    self.backgroundColor = [UIColor whiteColor];
+  }
+  return self;
+}
+
+- (void)setIsAudioLoopPlaying:(BOOL)isAudioLoopPlaying {
+  if (_isAudioLoopPlaying == isAudioLoopPlaying) {
+    return;
+  }
+  _isAudioLoopPlaying = isAudioLoopPlaying;
+  [self updateAudioLoopButton];
+}
+
+- (void)layoutSubviews {
+  CGRect bounds = self.bounds;
+  CGFloat roomTextWidth = bounds.size.width - 2 * kRoomTextFieldMargin;
+  CGFloat roomTextHeight = [_roomText sizeThatFits:bounds.size].height;
+  _roomText.frame =
+      CGRectMake(kRoomTextFieldMargin, kRoomTextFieldMargin, roomTextWidth,
+                 roomTextHeight);
+
+  CGFloat buttonHeight =
+      (CGRectGetMaxY(self.bounds) - CGRectGetMaxY(_roomText.frame) - kCallControlMargin * 4) / 3;
+
+  CGFloat regularCallFrameTop = CGRectGetMaxY(_roomText.frame) + kCallControlMargin;
+  CGRect regularCallFrame = CGRectMake(kCallControlMargin,
+                                       regularCallFrameTop,
+                                       bounds.size.width - 2*kCallControlMargin,
+                                       buttonHeight);
+
+  CGFloat loopbackCallFrameTop = CGRectGetMaxY(regularCallFrame) + kCallControlMargin;
+  CGRect loopbackCallFrame = CGRectMake(kCallControlMargin,
+                                        loopbackCallFrameTop,
+                                        bounds.size.width - 2*kCallControlMargin,
+                                        buttonHeight);
+
+  CGFloat audioLoopTop = CGRectGetMaxY(loopbackCallFrame) + kCallControlMargin;
+  CGRect audioLoopFrame = CGRectMake(kCallControlMargin,
+                                     audioLoopTop,
+                                     bounds.size.width - 2*kCallControlMargin,
+                                     buttonHeight);
+
+  _startRegularCallButton.frame = regularCallFrame;
+  _startLoopbackCallButton.frame = loopbackCallFrame;
+  _audioLoopButton.frame = audioLoopFrame;
+}
+
+#pragma mark - Private
+
+- (void)updateAudioLoopButton {
+  if (_isAudioLoopPlaying) {
+    [_audioLoopButton setTitle:@"Stop sound" forState:UIControlStateNormal];
+  } else {
+    [_audioLoopButton setTitle:@"Play sound" forState:UIControlStateNormal];
+  }
+}
+
+- (void)onToggleAudioLoop:(id)sender {
+  [_delegate mainViewDidToggleAudioLoop:self];
+}
+
+- (void)onStartRegularCall:(id)sender {
+  [_delegate mainView:self didInputRoom:_roomText.roomText isLoopback:NO];
+}
+
+- (void)onStartLoopbackCall:(id)sender {
+  [_delegate mainView:self didInputRoom:_roomText.roomText isLoopback:YES];
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.h
new file mode 100644
index 0000000000..e5c92dd304
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.h
@@ -0,0 +1,14 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+@interface ARDMainViewController : UIViewController
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m
new file mode 100644
index 0000000000..e8b8112e41
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m
@@ -0,0 +1,263 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDMainViewController.h"
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "sdk/objc/base/RTCLogging.h"
+#import "sdk/objc/components/audio/RTCAudioSession.h"
+#import "sdk/objc/components/audio/RTCAudioSessionConfiguration.h"
+#import "sdk/objc/helpers/RTCDispatcher.h"
+
+#import "ARDAppClient.h"
+#import "ARDMainView.h"
+#import "ARDSettingsModel.h"
+#import "ARDSettingsViewController.h"
+#import "ARDVideoCallViewController.h"
+
+static NSString *const barButtonImageString = @"ic_settings_black_24dp.png";
+
+// Launch argument to be passed to indicate that the app should start loopback immediatly
+static NSString *const loopbackLaunchProcessArgument = @"loopback";
+
+@interface ARDMainViewController () <ARDMainViewDelegate,
+                                     ARDVideoCallViewControllerDelegate,
+                                     RTC_OBJC_TYPE (RTCAudioSessionDelegate)>
+@property(nonatomic, strong) ARDMainView *mainView;
+@property(nonatomic, strong) AVAudioPlayer *audioPlayer;
+@end
+
+@implementation ARDMainViewController {
+  BOOL _useManualAudio;
+}
+
+@synthesize mainView = _mainView;
+@synthesize audioPlayer = _audioPlayer;
+
+- (void)viewDidLoad {
+  [super viewDidLoad];
+  if ([[[NSProcessInfo processInfo] arguments] containsObject:loopbackLaunchProcessArgument]) {
+    [self mainView:nil didInputRoom:@"" isLoopback:YES];
+  }
+}
+
+- (void)loadView {
+  self.title = @"AppRTC Mobile";
+  _mainView = [[ARDMainView alloc] initWithFrame:CGRectZero];
+  _mainView.delegate = self;
+  self.view = _mainView;
+  [self addSettingsBarButton];
+
+  RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *webRTCConfig =
+      [RTC_OBJC_TYPE(RTCAudioSessionConfiguration) webRTCConfiguration];
+  webRTCConfig.categoryOptions = webRTCConfig.categoryOptions |
+      AVAudioSessionCategoryOptionDefaultToSpeaker;
+  [RTC_OBJC_TYPE(RTCAudioSessionConfiguration) setWebRTCConfiguration:webRTCConfig];
+
+  RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  [session addDelegate:self];
+
+  [self configureAudioSession];
+  [self setupAudioPlayer];
+}
+
+- (void)addSettingsBarButton {
+  UIBarButtonItem *settingsButton =
+      [[UIBarButtonItem alloc] initWithImage:[UIImage imageNamed:barButtonImageString]
+                                       style:UIBarButtonItemStylePlain
+                                      target:self
+                                      action:@selector(showSettings:)];
+  self.navigationItem.rightBarButtonItem = settingsButton;
+}
+
++ (NSString *)loopbackRoomString {
+  NSString *loopbackRoomString =
+      [[NSUUID UUID].UUIDString stringByReplacingOccurrencesOfString:@"-" withString:@""];
+  return loopbackRoomString;
+}
+
+#pragma mark - ARDMainViewDelegate
+
+- (void)mainView:(ARDMainView *)mainView didInputRoom:(NSString *)room isLoopback:(BOOL)isLoopback {
+  if (!room.length) {
+    if (isLoopback) {
+      // If this is a loopback call, allow a generated room name.
+      room = [[self class] loopbackRoomString];
+    } else {
+      [self showAlertWithMessage:@"Missing room name."];
+      return;
+    }
+  }
+  // Trim whitespaces.
+  NSCharacterSet *whitespaceSet = [NSCharacterSet whitespaceCharacterSet];
+  NSString *trimmedRoom = [room stringByTrimmingCharactersInSet:whitespaceSet];
+
+  // Check that room name is valid.
+  NSError *error = nil;
+  NSRegularExpressionOptions options = NSRegularExpressionCaseInsensitive;
+  NSRegularExpression *regex =
+      [NSRegularExpression regularExpressionWithPattern:@"\\w+"
+                                                options:options
+                                                  error:&error];
+  if (error) {
+    [self showAlertWithMessage:error.localizedDescription];
+    return;
+  }
+  NSRange matchRange =
+      [regex rangeOfFirstMatchInString:trimmedRoom
+                               options:0
+                                 range:NSMakeRange(0, trimmedRoom.length)];
+  if (matchRange.location == NSNotFound ||
+      matchRange.length != trimmedRoom.length) {
+    [self showAlertWithMessage:@"Invalid room name."];
+    return;
+  }
+
+  ARDSettingsModel *settingsModel = [[ARDSettingsModel alloc] init];
+
+  RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  session.useManualAudio = [settingsModel currentUseManualAudioConfigSettingFromStore];
+  session.isAudioEnabled = NO;
+
+  // Kick off the video call.
+  ARDVideoCallViewController *videoCallViewController =
+      [[ARDVideoCallViewController alloc] initForRoom:trimmedRoom
+                                           isLoopback:isLoopback
+                                             delegate:self];
+  videoCallViewController.modalTransitionStyle =
+      UIModalTransitionStyleCrossDissolve;
+  videoCallViewController.modalPresentationStyle = UIModalPresentationFullScreen;
+  [self presentViewController:videoCallViewController
+                     animated:YES
+                   completion:nil];
+}
+
+- (void)mainViewDidToggleAudioLoop:(ARDMainView *)mainView {
+  if (mainView.isAudioLoopPlaying) {
+    [_audioPlayer stop];
+  } else {
+    [_audioPlayer play];
+  }
+  mainView.isAudioLoopPlaying = _audioPlayer.playing;
+}
+
+#pragma mark - ARDVideoCallViewControllerDelegate
+
+- (void)viewControllerDidFinish:(ARDVideoCallViewController *)viewController {
+  if (![viewController isBeingDismissed]) {
+    RTCLog(@"Dismissing VC");
+    [self dismissViewControllerAnimated:YES completion:^{
+      [self restartAudioPlayerIfNeeded];
+    }];
+  }
+  RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  session.isAudioEnabled = NO;
+}
+
+#pragma mark - RTC_OBJC_TYPE(RTCAudioSessionDelegate)
+
+- (void)audioSessionDidStartPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+  // Stop playback on main queue and then configure WebRTC.
+  [RTC_OBJC_TYPE(RTCDispatcher)
+      dispatchAsyncOnType:RTCDispatcherTypeMain
+                    block:^{
+                      if (self.mainView.isAudioLoopPlaying) {
+                        RTCLog(@"Stopping audio loop due to WebRTC start.");
+                        [self.audioPlayer stop];
+                      }
+                      RTCLog(@"Setting isAudioEnabled to YES.");
+                      session.isAudioEnabled = YES;
+                    }];
+}
+
+- (void)audioSessionDidStopPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+  // WebRTC is done with the audio session. Restart playback.
+  [RTC_OBJC_TYPE(RTCDispatcher) dispatchAsyncOnType:RTCDispatcherTypeMain
+                                              block:^{
+                                                RTCLog(@"audioSessionDidStopPlayOrRecord");
+                                                [self restartAudioPlayerIfNeeded];
+                                              }];
+}
+
+#pragma mark - Private
+- (void)showSettings:(id)sender {
+  ARDSettingsViewController *settingsController =
+      [[ARDSettingsViewController alloc] initWithStyle:UITableViewStyleGrouped
+                                         settingsModel:[[ARDSettingsModel alloc] init]];
+
+  UINavigationController *navigationController =
+      [[UINavigationController alloc] initWithRootViewController:settingsController];
+  [self presentViewControllerAsModal:navigationController];
+}
+
+- (void)presentViewControllerAsModal:(UIViewController *)viewController {
+  [self presentViewController:viewController animated:YES completion:nil];
+}
+
+- (void)configureAudioSession {
+  RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *configuration =
+      [[RTC_OBJC_TYPE(RTCAudioSessionConfiguration) alloc] init];
+  configuration.category = AVAudioSessionCategoryAmbient;
+  configuration.categoryOptions = AVAudioSessionCategoryOptionDuckOthers;
+  configuration.mode = AVAudioSessionModeDefault;
+
+  RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  [session lockForConfiguration];
+  BOOL hasSucceeded = NO;
+  NSError *error = nil;
+  if (session.isActive) {
+    hasSucceeded = [session setConfiguration:configuration error:&error];
+  } else {
+    hasSucceeded = [session setConfiguration:configuration
+                                      active:YES
+                                       error:&error];
+  }
+  if (!hasSucceeded) {
+    RTCLogError(@"Error setting configuration: %@", error.localizedDescription);
+  }
+  [session unlockForConfiguration];
+}
+
+- (void)setupAudioPlayer {
+  NSString *audioFilePath =
+      [[NSBundle mainBundle] pathForResource:@"mozart" ofType:@"mp3"];
+  NSURL *audioFileURL = [NSURL URLWithString:audioFilePath];
+  _audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:audioFileURL
+                                                        error:nil];
+  _audioPlayer.numberOfLoops = -1;
+  _audioPlayer.volume = 1.0;
+  [_audioPlayer prepareToPlay];
+}
+
+- (void)restartAudioPlayerIfNeeded {
+  [self configureAudioSession];
+  if (_mainView.isAudioLoopPlaying && !self.presentedViewController) {
+    RTCLog(@"Starting audio loop due to WebRTC end.");
+    [_audioPlayer play];
+  }
+}
+
+- (void)showAlertWithMessage:(NSString*)message {
+  UIAlertController *alert =
+      [UIAlertController alertControllerWithTitle:nil
+                                          message:message
+                                   preferredStyle:UIAlertControllerStyleAlert];
+
+  UIAlertAction *defaultAction = [UIAlertAction actionWithTitle:@"OK"
+                                                          style:UIAlertActionStyleDefault
+                                                        handler:^(UIAlertAction *action){
+                                                        }];
+
+  [alert addAction:defaultAction];
+  [self presentViewController:alert animated:YES completion:nil];
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.h
new file mode 100644
index 0000000000..759af5416f
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+@class ARDSettingsModel;
+
+NS_ASSUME_NONNULL_BEGIN
+/**
+ * Displays settings options.
+ */
+@interface ARDSettingsViewController : UITableViewController
+
+/**
+ * Creates new instance.
+ *
+ * @param style the table view style that should be used
+ * @param settingsModel model class for the user settings.
+ */
+- (instancetype)initWithStyle:(UITableViewStyle)style
+                settingsModel:(ARDSettingsModel *)settingsModel;
+
+#pragma mark - Unavailable
+
+- (instancetype)initWithStyle:(UITableViewStyle)style NS_UNAVAILABLE;
+- (instancetype)init NS_UNAVAILABLE;
++ (instancetype)new NS_UNAVAILABLE;
+
+@end
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.m
new file mode 100644
index 0000000000..9bcbd3aa5c
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.m
@@ -0,0 +1,361 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDSettingsViewController.h"
+#import "ARDSettingsModel.h"
+#import "RTCVideoCodecInfo+HumanReadable.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef NS_ENUM(int, ARDSettingsSections) {
+  ARDSettingsSectionAudioSettings = 0,
+  ARDSettingsSectionVideoResolution,
+  ARDSettingsSectionVideoCodec,
+  ARDSettingsSectionBitRate,
+};
+
+typedef NS_ENUM(int, ARDAudioSettingsOptions) {
+  ARDAudioSettingsAudioOnly = 0,
+  ARDAudioSettingsCreateAecDump,
+  ARDAudioSettingsUseManualAudioConfig,
+};
+
+@interface ARDSettingsViewController () <UITextFieldDelegate> {
+  ARDSettingsModel *_settingsModel;
+}
+
+@end
+
+@implementation ARDSettingsViewController
+
+- (instancetype)initWithStyle:(UITableViewStyle)style
+                settingsModel:(ARDSettingsModel *)settingsModel {
+  self = [super initWithStyle:style];
+  if (self) {
+    _settingsModel = settingsModel;
+  }
+  return self;
+}
+
+#pragma mark - View lifecycle
+
+- (void)viewDidLoad {
+  [super viewDidLoad];
+  self.title = @"Settings";
+  [self addDoneBarButton];
+}
+
+- (void)viewWillAppear:(BOOL)animated {
+  [super viewWillAppear:animated];
+}
+
+#pragma mark - Data source
+
+- (NSArray<NSString *> *)videoResolutionArray {
+  return [_settingsModel availableVideoResolutions];
+}
+
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)videoCodecArray {
+  return [_settingsModel availableVideoCodecs];
+}
+
+#pragma mark -
+
+- (void)addDoneBarButton {
+  UIBarButtonItem *barItem =
+      [[UIBarButtonItem alloc] initWithBarButtonSystemItem:UIBarButtonSystemItemDone
+                                                    target:self
+                                                    action:@selector(dismissModally:)];
+  self.navigationItem.leftBarButtonItem = barItem;
+}
+
+#pragma mark - Dismissal of view controller
+
+- (void)dismissModally:(id)sender {
+  [self dismissViewControllerAnimated:YES completion:nil];
+}
+
+#pragma mark - Table view data source
+
+- (NSInteger)numberOfSectionsInTableView:(UITableView *)tableView {
+  return 4;
+}
+
+- (NSInteger)tableView:(UITableView *)tableView numberOfRowsInSection:(NSInteger)section {
+  switch (section) {
+    case ARDSettingsSectionAudioSettings:
+      return 3;
+    case ARDSettingsSectionVideoResolution:
+      return self.videoResolutionArray.count;
+    case ARDSettingsSectionVideoCodec:
+      return self.videoCodecArray.count;
+    default:
+      return 1;
+  }
+}
+
+#pragma mark - Table view delegate helpers
+
+- (void)removeAllAccessories:(UITableView *)tableView
+                   inSection:(int)section
+{
+  for (int i = 0; i < [tableView numberOfRowsInSection:section]; i++) {
+    NSIndexPath *rowPath = [NSIndexPath indexPathForRow:i inSection:section];
+    UITableViewCell *cell = [tableView cellForRowAtIndexPath:rowPath];
+    cell.accessoryType = UITableViewCellAccessoryNone;
+  }
+}
+
+- (void)tableView:(UITableView *)tableView
+updateListSelectionAtIndexPath:(NSIndexPath *)indexPath
+        inSection:(int)section {
+  [self removeAllAccessories:tableView inSection:section];
+  UITableViewCell *cell = [tableView cellForRowAtIndexPath:indexPath];
+  cell.accessoryType = UITableViewCellAccessoryCheckmark;
+  [tableView deselectRowAtIndexPath:indexPath animated:YES];
+}
+
+#pragma mark - Table view delegate
+
+- (nullable NSString *)tableView:(UITableView *)tableView
+         titleForHeaderInSection:(NSInteger)section {
+  switch (section) {
+    case ARDSettingsSectionAudioSettings:
+      return @"Audio";
+    case ARDSettingsSectionVideoResolution:
+      return @"Video resolution";
+    case ARDSettingsSectionVideoCodec:
+      return @"Video codec";
+    case ARDSettingsSectionBitRate:
+      return @"Maximum bitrate";
+    default:
+      return @"";
+  }
+}
+
+- (UITableViewCell *)tableView:(UITableView *)tableView
+         cellForRowAtIndexPath:(NSIndexPath *)indexPath {
+  switch (indexPath.section) {
+    case ARDSettingsSectionAudioSettings:
+      return [self audioSettingsTableViewCellForTableView:tableView atIndexPath:indexPath];
+
+    case ARDSettingsSectionVideoResolution:
+      return [self videoResolutionTableViewCellForTableView:tableView atIndexPath:indexPath];
+
+    case ARDSettingsSectionVideoCodec:
+      return [self videoCodecTableViewCellForTableView:tableView atIndexPath:indexPath];
+
+    case ARDSettingsSectionBitRate:
+      return [self bitrateTableViewCellForTableView:tableView atIndexPath:indexPath];
+
+    default:
+      return [[UITableViewCell alloc] initWithStyle:UITableViewCellStyleDefault
+                                    reuseIdentifier:@"identifier"];
+  }
+}
+
+- (void)tableView:(UITableView *)tableView didSelectRowAtIndexPath:(NSIndexPath *)indexPath {
+  switch (indexPath.section) {
+    case ARDSettingsSectionVideoResolution:
+      [self tableView:tableView disSelectVideoResolutionAtIndex:indexPath];
+      break;
+
+    case ARDSettingsSectionVideoCodec:
+      [self tableView:tableView didSelectVideoCodecCellAtIndexPath:indexPath];
+      break;
+  }
+}
+
+#pragma mark - Table view delegate(Video Resolution)
+
+- (UITableViewCell *)videoResolutionTableViewCellForTableView:(UITableView *)tableView
+                                                  atIndexPath:(NSIndexPath *)indexPath {
+  NSString *dequeueIdentifier = @"ARDSettingsVideoResolutionViewCellIdentifier";
+  UITableViewCell *cell = [tableView dequeueReusableCellWithIdentifier:dequeueIdentifier];
+  if (!cell) {
+    cell = [[UITableViewCell alloc] initWithStyle:UITableViewCellStyleDefault
+                                  reuseIdentifier:dequeueIdentifier];
+  }
+  NSString *resolution = self.videoResolutionArray[indexPath.row];
+  cell.textLabel.text = resolution;
+  if ([resolution isEqualToString:[_settingsModel currentVideoResolutionSettingFromStore]]) {
+    cell.accessoryType = UITableViewCellAccessoryCheckmark;
+  } else {
+    cell.accessoryType = UITableViewCellAccessoryNone;
+  }
+
+  return cell;
+}
+
+- (void)tableView:(UITableView *)tableView
+    disSelectVideoResolutionAtIndex:(NSIndexPath *)indexPath {
+  [self tableView:tableView
+      updateListSelectionAtIndexPath:indexPath
+                           inSection:ARDSettingsSectionVideoResolution];
+
+  NSString *videoResolution = self.videoResolutionArray[indexPath.row];
+  [_settingsModel storeVideoResolutionSetting:videoResolution];
+}
+
+#pragma mark - Table view delegate(Video Codec)
+
+- (UITableViewCell *)videoCodecTableViewCellForTableView:(UITableView *)tableView
+                                             atIndexPath:(NSIndexPath *)indexPath {
+  NSString *dequeueIdentifier = @"ARDSettingsVideoCodecCellIdentifier";
+  UITableViewCell *cell = [tableView dequeueReusableCellWithIdentifier:dequeueIdentifier];
+  if (!cell) {
+    cell = [[UITableViewCell alloc] initWithStyle:UITableViewCellStyleDefault
+                                  reuseIdentifier:dequeueIdentifier];
+  }
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *codec = self.videoCodecArray[indexPath.row];
+  cell.textLabel.text = [codec humanReadableDescription];
+  if ([codec isEqualToCodecInfo:[_settingsModel currentVideoCodecSettingFromStore]]) {
+    cell.accessoryType = UITableViewCellAccessoryCheckmark;
+  } else {
+    cell.accessoryType = UITableViewCellAccessoryNone;
+  }
+
+  return cell;
+}
+
+- (void)tableView:(UITableView *)tableView
+    didSelectVideoCodecCellAtIndexPath:(NSIndexPath *)indexPath {
+  [self tableView:tableView
+    updateListSelectionAtIndexPath:indexPath
+        inSection:ARDSettingsSectionVideoCodec];
+
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *videoCodec = self.videoCodecArray[indexPath.row];
+  [_settingsModel storeVideoCodecSetting:videoCodec];
+}
+
+#pragma mark - Table view delegate(Bitrate)
+
+- (UITableViewCell *)bitrateTableViewCellForTableView:(UITableView *)tableView
+                                          atIndexPath:(NSIndexPath *)indexPath {
+  NSString *dequeueIdentifier = @"ARDSettingsBitrateCellIdentifier";
+  UITableViewCell *cell = [tableView dequeueReusableCellWithIdentifier:dequeueIdentifier];
+  if (!cell) {
+    cell = [[UITableViewCell alloc] initWithStyle:UITableViewCellStyleDefault
+                                  reuseIdentifier:dequeueIdentifier];
+
+    UITextField *textField = [[UITextField alloc]
+        initWithFrame:CGRectMake(10, 0, cell.bounds.size.width - 20, cell.bounds.size.height)];
+    NSString *currentMaxBitrate = [_settingsModel currentMaxBitrateSettingFromStore].stringValue;
+    textField.text = currentMaxBitrate;
+    textField.placeholder = @"Enter max bit rate (kbps)";
+    textField.keyboardType = UIKeyboardTypeNumberPad;
+    textField.delegate = self;
+
+    // Numerical keyboards have no return button, we need to add one manually.
+    UIToolbar *numberToolbar =
+        [[UIToolbar alloc] initWithFrame:CGRectMake(0, 0, self.view.bounds.size.width, 50)];
+    numberToolbar.items = @[
+      [[UIBarButtonItem alloc] initWithBarButtonSystemItem:UIBarButtonSystemItemFlexibleSpace
+                                                    target:nil
+                                                    action:nil],
+      [[UIBarButtonItem alloc] initWithTitle:@"Apply"
+                                       style:UIBarButtonItemStyleDone
+                                      target:self
+                                      action:@selector(numberTextFieldDidEndEditing:)]
+    ];
+    [numberToolbar sizeToFit];
+
+    textField.inputAccessoryView = numberToolbar;
+    [cell addSubview:textField];
+  }
+  return cell;
+}
+
+- (void)numberTextFieldDidEndEditing:(id)sender {
+  [self.view endEditing:YES];
+}
+
+- (void)textFieldDidEndEditing:(UITextField *)textField {
+  NSNumber *bitrateNumber = nil;
+
+  if (textField.text.length != 0) {
+    bitrateNumber = [NSNumber numberWithInteger:textField.text.intValue];
+  }
+
+  [_settingsModel storeMaxBitrateSetting:bitrateNumber];
+}
+
+#pragma mark - Table view delegate(Audio settings)
+
+- (UITableViewCell *)audioSettingsTableViewCellForTableView:(UITableView *)tableView
+                                                atIndexPath:(NSIndexPath *)indexPath {
+  NSString *dequeueIdentifier = @"ARDSettingsAudioSettingsCellIdentifier";
+  UITableViewCell *cell = [tableView dequeueReusableCellWithIdentifier:dequeueIdentifier];
+  if (!cell) {
+    cell = [[UITableViewCell alloc] initWithStyle:UITableViewCellStyleDefault
+                                  reuseIdentifier:dequeueIdentifier];
+    cell.selectionStyle = UITableViewCellSelectionStyleNone;
+    UISwitch *switchView = [[UISwitch alloc] initWithFrame:CGRectZero];
+    switchView.tag = indexPath.row;
+    [switchView addTarget:self
+                   action:@selector(audioSettingSwitchChanged:)
+         forControlEvents:UIControlEventValueChanged];
+    cell.accessoryView = switchView;
+  }
+
+  cell.textLabel.text = [self labelForAudioSettingAtIndexPathRow:indexPath.row];
+  UISwitch *switchView = (UISwitch *)cell.accessoryView;
+  switchView.on = [self valueForAudioSettingAtIndexPathRow:indexPath.row];
+
+  return cell;
+}
+
+- (NSString *)labelForAudioSettingAtIndexPathRow:(NSInteger)setting {
+  switch (setting) {
+    case ARDAudioSettingsAudioOnly:
+      return @"Audio only";
+    case ARDAudioSettingsCreateAecDump:
+      return @"Create AecDump";
+    case ARDAudioSettingsUseManualAudioConfig:
+      return @"Use manual audio config";
+    default:
+      return @"";
+  }
+}
+
+- (BOOL)valueForAudioSettingAtIndexPathRow:(NSInteger)setting {
+  switch (setting) {
+    case ARDAudioSettingsAudioOnly:
+      return [_settingsModel currentAudioOnlySettingFromStore];
+    case ARDAudioSettingsCreateAecDump:
+      return [_settingsModel currentCreateAecDumpSettingFromStore];
+    case ARDAudioSettingsUseManualAudioConfig:
+      return [_settingsModel currentUseManualAudioConfigSettingFromStore];
+    default:
+      return NO;
+  }
+}
+
+- (void)audioSettingSwitchChanged:(UISwitch *)sender {
+  switch (sender.tag) {
+    case ARDAudioSettingsAudioOnly: {
+      [_settingsModel storeAudioOnlySetting:sender.isOn];
+      break;
+    }
+    case ARDAudioSettingsCreateAecDump: {
+      [_settingsModel storeCreateAecDumpSetting:sender.isOn];
+      break;
+    }
+    case ARDAudioSettingsUseManualAudioConfig: {
+      [_settingsModel storeUseManualAudioConfigSetting:sender.isOn];
+      break;
+    }
+    default:
+      break;
+  }
+}
+
+@end
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDStatsView.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDStatsView.h
new file mode 100644
index 0000000000..72207de64e
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDStatsView.h
@@ -0,0 +1,21 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+#import "sdk/objc/base/RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCStatisticsReport);
+
+@interface ARDStatsView : UIView
+
+- (void)setStats:(RTC_OBJC_TYPE(RTCStatisticsReport) *)stats;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDStatsView.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDStatsView.m
new file mode 100644
index 0000000000..867ba5b09e
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDStatsView.m
@@ -0,0 +1,50 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDStatsView.h"
+
+#import "sdk/objc/api/peerconnection/RTCLegacyStatsReport.h"
+
+#import "ARDStatsBuilder.h"
+
+@implementation ARDStatsView {
+  UILabel *_statsLabel;
+  ARDStatsBuilder *_statsBuilder;
+}
+
+- (instancetype)initWithFrame:(CGRect)frame {
+  if (self = [super initWithFrame:frame]) {
+    _statsLabel = [[UILabel alloc] initWithFrame:CGRectZero];
+    _statsLabel.numberOfLines = 0;
+    _statsLabel.font = [UIFont fontWithName:@"Roboto" size:12];
+    _statsLabel.adjustsFontSizeToFitWidth = YES;
+    _statsLabel.minimumScaleFactor = 0.6;
+    _statsLabel.textColor = [UIColor greenColor];
+    [self addSubview:_statsLabel];
+    self.backgroundColor = [UIColor colorWithWhite:0 alpha:.6];
+    _statsBuilder = [[ARDStatsBuilder alloc] init];
+  }
+  return self;
+}
+
+- (void)setStats:(RTC_OBJC_TYPE(RTCStatisticsReport) *)stats {
+  _statsBuilder.stats = stats;
+  _statsLabel.text = _statsBuilder.statsString;
+}
+
+- (void)layoutSubviews {
+  _statsLabel.frame = self.bounds;
+}
+
+- (CGSize)sizeThatFits:(CGSize)size {
+  return [_statsLabel sizeThatFits:size];
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallView.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallView.h
new file mode 100644
index 0000000000..a31c7fe742
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallView.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+#import "sdk/objc/base/RTCVideoRenderer.h"
+#import "sdk/objc/helpers/RTCCameraPreviewView.h"
+
+#import "ARDStatsView.h"
+
+@class ARDVideoCallView;
+@protocol ARDVideoCallViewDelegate <NSObject>
+
+// Called when the camera switch button is pressed.
+- (void)videoCallView:(ARDVideoCallView *)view
+    shouldSwitchCameraWithCompletion:(void (^)(NSError *))completion;
+
+// Called when the route change button is pressed.
+- (void)videoCallView:(ARDVideoCallView *)view
+    shouldChangeRouteWithCompletion:(void (^)(void))completion;
+
+// Called when the hangup button is pressed.
+- (void)videoCallViewDidHangup:(ARDVideoCallView *)view;
+
+// Called when stats are enabled by triple tapping.
+- (void)videoCallViewDidEnableStats:(ARDVideoCallView *)view;
+
+@end
+
+// Video call view that shows local and remote video, provides a label to
+// display status, and also a hangup button.
+@interface ARDVideoCallView : UIView
+
+@property(nonatomic, readonly) UILabel *statusLabel;
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCCameraPreviewView) * localVideoView;
+@property(nonatomic, readonly) __kindof UIView<RTC_OBJC_TYPE(RTCVideoRenderer)> *remoteVideoView;
+@property(nonatomic, readonly) ARDStatsView *statsView;
+@property(nonatomic, weak) id<ARDVideoCallViewDelegate> delegate;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallView.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallView.m
new file mode 100644
index 0000000000..437aea8d56
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallView.m
@@ -0,0 +1,213 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDVideoCallView.h"
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "sdk/objc/components/renderer/metal/RTCMTLVideoView.h"
+
+#import "UIImage+ARDUtilities.h"
+
+static CGFloat const kButtonPadding = 16;
+static CGFloat const kButtonSize = 48;
+static CGFloat const kLocalVideoViewSize = 120;
+static CGFloat const kLocalVideoViewPadding = 8;
+static CGFloat const kStatusBarHeight = 20;
+
+@interface ARDVideoCallView () <RTC_OBJC_TYPE (RTCVideoViewDelegate)>
+@end
+
+@implementation ARDVideoCallView {
+  UIButton *_routeChangeButton;
+  UIButton *_cameraSwitchButton;
+  UIButton *_hangupButton;
+  CGSize _remoteVideoSize;
+}
+
+@synthesize statusLabel = _statusLabel;
+@synthesize localVideoView = _localVideoView;
+@synthesize remoteVideoView = _remoteVideoView;
+@synthesize statsView = _statsView;
+@synthesize delegate = _delegate;
+
+- (instancetype)initWithFrame:(CGRect)frame {
+  if (self = [super initWithFrame:frame]) {
+
+    _remoteVideoView = [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectZero];
+
+    [self addSubview:_remoteVideoView];
+
+    _localVideoView = [[RTC_OBJC_TYPE(RTCCameraPreviewView) alloc] initWithFrame:CGRectZero];
+    [self addSubview:_localVideoView];
+
+    _statsView = [[ARDStatsView alloc] initWithFrame:CGRectZero];
+    _statsView.hidden = YES;
+    [self addSubview:_statsView];
+
+    _routeChangeButton = [UIButton buttonWithType:UIButtonTypeCustom];
+    _routeChangeButton.backgroundColor = [UIColor grayColor];
+    _routeChangeButton.layer.cornerRadius = kButtonSize / 2;
+    _routeChangeButton.layer.masksToBounds = YES;
+    UIImage *image = [UIImage imageForName:@"ic_surround_sound_black_24dp.png"
+                                     color:[UIColor whiteColor]];
+    [_routeChangeButton setImage:image forState:UIControlStateNormal];
+    [_routeChangeButton addTarget:self
+                           action:@selector(onRouteChange:)
+                 forControlEvents:UIControlEventTouchUpInside];
+    [self addSubview:_routeChangeButton];
+
+    // TODO(tkchin): don't display this if we can't actually do camera switch.
+    _cameraSwitchButton = [UIButton buttonWithType:UIButtonTypeCustom];
+    _cameraSwitchButton.backgroundColor = [UIColor grayColor];
+    _cameraSwitchButton.layer.cornerRadius = kButtonSize / 2;
+    _cameraSwitchButton.layer.masksToBounds = YES;
+    image = [UIImage imageForName:@"ic_switch_video_black_24dp.png" color:[UIColor whiteColor]];
+    [_cameraSwitchButton setImage:image forState:UIControlStateNormal];
+    [_cameraSwitchButton addTarget:self
+                      action:@selector(onCameraSwitch:)
+            forControlEvents:UIControlEventTouchUpInside];
+    [self addSubview:_cameraSwitchButton];
+
+    _hangupButton = [UIButton buttonWithType:UIButtonTypeCustom];
+    _hangupButton.backgroundColor = [UIColor redColor];
+    _hangupButton.layer.cornerRadius = kButtonSize / 2;
+    _hangupButton.layer.masksToBounds = YES;
+    image = [UIImage imageForName:@"ic_call_end_black_24dp.png"
+                            color:[UIColor whiteColor]];
+    [_hangupButton setImage:image forState:UIControlStateNormal];
+    [_hangupButton addTarget:self
+                      action:@selector(onHangup:)
+            forControlEvents:UIControlEventTouchUpInside];
+    [self addSubview:_hangupButton];
+
+    _statusLabel = [[UILabel alloc] initWithFrame:CGRectZero];
+    _statusLabel.font = [UIFont fontWithName:@"Roboto" size:16];
+    _statusLabel.textColor = [UIColor whiteColor];
+    [self addSubview:_statusLabel];
+
+    UITapGestureRecognizer *tapRecognizer =
+        [[UITapGestureRecognizer alloc]
+            initWithTarget:self
+                    action:@selector(didTripleTap:)];
+    tapRecognizer.numberOfTapsRequired = 3;
+    [self addGestureRecognizer:tapRecognizer];
+  }
+  return self;
+}
+
+- (void)layoutSubviews {
+  CGRect bounds = self.bounds;
+  if (_remoteVideoSize.width > 0 && _remoteVideoSize.height > 0) {
+    // Aspect fill remote video into bounds.
+    CGRect remoteVideoFrame =
+        AVMakeRectWithAspectRatioInsideRect(_remoteVideoSize, bounds);
+    CGFloat scale = 1;
+    if (remoteVideoFrame.size.width > remoteVideoFrame.size.height) {
+      // Scale by height.
+      scale = bounds.size.height / remoteVideoFrame.size.height;
+    } else {
+      // Scale by width.
+      scale = bounds.size.width / remoteVideoFrame.size.width;
+    }
+    remoteVideoFrame.size.height *= scale;
+    remoteVideoFrame.size.width *= scale;
+    _remoteVideoView.frame = remoteVideoFrame;
+    _remoteVideoView.center =
+        CGPointMake(CGRectGetMidX(bounds), CGRectGetMidY(bounds));
+  } else {
+    _remoteVideoView.frame = bounds;
+  }
+
+  // Aspect fit local video view into a square box.
+  CGRect localVideoFrame =
+      CGRectMake(0, 0, kLocalVideoViewSize, kLocalVideoViewSize);
+  // Place the view in the bottom right.
+  localVideoFrame.origin.x = CGRectGetMaxX(bounds)
+      - localVideoFrame.size.width - kLocalVideoViewPadding;
+  localVideoFrame.origin.y = CGRectGetMaxY(bounds)
+      - localVideoFrame.size.height - kLocalVideoViewPadding;
+  _localVideoView.frame = localVideoFrame;
+
+  // Place stats at the top.
+  CGSize statsSize = [_statsView sizeThatFits:bounds.size];
+  _statsView.frame = CGRectMake(CGRectGetMinX(bounds),
+                                CGRectGetMinY(bounds) + kStatusBarHeight,
+                                statsSize.width, statsSize.height);
+
+  // Place hangup button in the bottom left.
+  _hangupButton.frame =
+      CGRectMake(CGRectGetMinX(bounds) + kButtonPadding,
+                 CGRectGetMaxY(bounds) - kButtonPadding -
+                     kButtonSize,
+                 kButtonSize,
+                 kButtonSize);
+
+  // Place button to the right of hangup button.
+  CGRect cameraSwitchFrame = _hangupButton.frame;
+  cameraSwitchFrame.origin.x =
+      CGRectGetMaxX(cameraSwitchFrame) + kButtonPadding;
+  _cameraSwitchButton.frame = cameraSwitchFrame;
+
+  // Place route button to the right of camera button.
+  CGRect routeChangeFrame = _cameraSwitchButton.frame;
+  routeChangeFrame.origin.x =
+      CGRectGetMaxX(routeChangeFrame) + kButtonPadding;
+  _routeChangeButton.frame = routeChangeFrame;
+
+  [_statusLabel sizeToFit];
+  _statusLabel.center =
+      CGPointMake(CGRectGetMidX(bounds), CGRectGetMidY(bounds));
+}
+
+#pragma mark - RTC_OBJC_TYPE(RTCVideoViewDelegate)
+
+- (void)videoView:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)videoView didChangeVideoSize:(CGSize)size {
+  if (videoView == _remoteVideoView) {
+    _remoteVideoSize = size;
+  }
+  [self setNeedsLayout];
+}
+
+#pragma mark - Private
+
+- (void)onCameraSwitch:(UIButton *)sender {
+  sender.enabled = false;
+  [_delegate videoCallView:self
+      shouldSwitchCameraWithCompletion:^(NSError *error) {
+        dispatch_async(dispatch_get_main_queue(), ^(void) {
+          sender.enabled = true;
+        });
+      }];
+}
+
+- (void)onRouteChange:(UIButton *)sender {
+  sender.enabled = false;
+  __weak ARDVideoCallView *weakSelf = self;
+  [_delegate videoCallView:self
+      shouldChangeRouteWithCompletion:^(void) {
+        ARDVideoCallView *strongSelf = weakSelf;
+        if (strongSelf) {
+          dispatch_async(dispatch_get_main_queue(), ^(void) {
+            sender.enabled = true;
+          });
+        }
+      }];
+}
+
+- (void)onHangup:(id)sender {
+  [_delegate videoCallViewDidHangup:self];
+}
+
+- (void)didTripleTap:(UITapGestureRecognizer *)recognizer {
+  [_delegate videoCallViewDidEnableStats:self];
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.h
new file mode 100644
index 0000000000..bdb8747524
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+@class ARDVideoCallViewController;
+@protocol ARDVideoCallViewControllerDelegate <NSObject>
+
+- (void)viewControllerDidFinish:(ARDVideoCallViewController *)viewController;
+
+@end
+
+@interface ARDVideoCallViewController : UIViewController
+
+@property(nonatomic, weak) id<ARDVideoCallViewControllerDelegate> delegate;
+
+- (instancetype)initForRoom:(NSString *)room
+                 isLoopback:(BOOL)isLoopback
+                   delegate:(id<ARDVideoCallViewControllerDelegate>)delegate;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m
new file mode 100644
index 0000000000..a82d90b290
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m
@@ -0,0 +1,250 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDVideoCallViewController.h"
+
+#import "sdk/objc/api/peerconnection/RTCMediaConstraints.h"
+#import "sdk/objc/base/RTCLogging.h"
+#import "sdk/objc/components/audio/RTCAudioSession.h"
+#import "sdk/objc/components/capturer/RTCCameraVideoCapturer.h"
+#import "sdk/objc/helpers/RTCDispatcher.h"
+
+#import "ARDAppClient.h"
+#import "ARDCaptureController.h"
+#import "ARDFileCaptureController.h"
+#import "ARDSettingsModel.h"
+#import "ARDVideoCallView.h"
+
+@interface ARDVideoCallViewController () <ARDAppClientDelegate,
+                                          ARDVideoCallViewDelegate,
+                                          RTC_OBJC_TYPE (RTCAudioSessionDelegate)>
+@property(nonatomic, strong) RTC_OBJC_TYPE(RTCVideoTrack) * remoteVideoTrack;
+@property(nonatomic, readonly) ARDVideoCallView *videoCallView;
+@property(nonatomic, assign) AVAudioSessionPortOverride portOverride;
+@end
+
+@implementation ARDVideoCallViewController {
+  ARDAppClient *_client;
+  RTC_OBJC_TYPE(RTCVideoTrack) * _remoteVideoTrack;
+  ARDCaptureController *_captureController;
+  ARDFileCaptureController *_fileCaptureController NS_AVAILABLE_IOS(10);
+}
+
+@synthesize videoCallView = _videoCallView;
+@synthesize remoteVideoTrack = _remoteVideoTrack;
+@synthesize delegate = _delegate;
+@synthesize portOverride = _portOverride;
+
+- (instancetype)initForRoom:(NSString *)room
+                 isLoopback:(BOOL)isLoopback
+                   delegate:(id<ARDVideoCallViewControllerDelegate>)delegate {
+  if (self = [super init]) {
+    ARDSettingsModel *settingsModel = [[ARDSettingsModel alloc] init];
+    _delegate = delegate;
+
+    _client = [[ARDAppClient alloc] initWithDelegate:self];
+    [_client connectToRoomWithId:room settings:settingsModel isLoopback:isLoopback];
+  }
+  return self;
+}
+
+- (void)loadView {
+  _videoCallView = [[ARDVideoCallView alloc] initWithFrame:CGRectZero];
+  _videoCallView.delegate = self;
+  _videoCallView.statusLabel.text =
+      [self statusTextForState:RTCIceConnectionStateNew];
+  self.view = _videoCallView;
+
+  RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  [session addDelegate:self];
+}
+
+- (UIInterfaceOrientationMask)supportedInterfaceOrientations {
+  return UIInterfaceOrientationMaskAll;
+}
+
+#pragma mark - ARDAppClientDelegate
+
+- (void)appClient:(ARDAppClient *)client
+    didChangeState:(ARDAppClientState)state {
+  switch (state) {
+    case kARDAppClientStateConnected:
+      RTCLog(@"Client connected.");
+      break;
+    case kARDAppClientStateConnecting:
+      RTCLog(@"Client connecting.");
+      break;
+    case kARDAppClientStateDisconnected:
+      RTCLog(@"Client disconnected.");
+      [self hangup];
+      break;
+  }
+}
+
+- (void)appClient:(ARDAppClient *)client
+    didChangeConnectionState:(RTCIceConnectionState)state {
+  RTCLog(@"ICE state changed: %ld", (long)state);
+  __weak ARDVideoCallViewController *weakSelf = self;
+  dispatch_async(dispatch_get_main_queue(), ^{
+    ARDVideoCallViewController *strongSelf = weakSelf;
+    strongSelf.videoCallView.statusLabel.text =
+        [strongSelf statusTextForState:state];
+  });
+}
+
+- (void)appClient:(ARDAppClient *)client
+    didCreateLocalCapturer:(RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)localCapturer {
+  _videoCallView.localVideoView.captureSession = localCapturer.captureSession;
+  ARDSettingsModel *settingsModel = [[ARDSettingsModel alloc] init];
+  _captureController =
+      [[ARDCaptureController alloc] initWithCapturer:localCapturer settings:settingsModel];
+  [_captureController startCapture];
+}
+
+- (void)appClient:(ARDAppClient *)client
+    didCreateLocalFileCapturer:(RTC_OBJC_TYPE(RTCFileVideoCapturer) *)fileCapturer {
+#if defined(__IPHONE_11_0) && (__IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_11_0)
+  if (@available(iOS 10, *)) {
+    _fileCaptureController = [[ARDFileCaptureController alloc] initWithCapturer:fileCapturer];
+    [_fileCaptureController startCapture];
+  }
+#endif
+}
+
+- (void)appClient:(ARDAppClient *)client
+    didReceiveLocalVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)localVideoTrack {
+}
+
+- (void)appClient:(ARDAppClient *)client
+    didReceiveRemoteVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)remoteVideoTrack {
+  self.remoteVideoTrack = remoteVideoTrack;
+  __weak ARDVideoCallViewController *weakSelf = self;
+  dispatch_async(dispatch_get_main_queue(), ^{
+    ARDVideoCallViewController *strongSelf = weakSelf;
+    strongSelf.videoCallView.statusLabel.hidden = YES;
+  });
+}
+
+- (void)appClient:(ARDAppClient *)client didGetStats:(RTC_OBJC_TYPE(RTCStatisticsReport) *)stats {
+  _videoCallView.statsView.stats = stats;
+  [_videoCallView setNeedsLayout];
+}
+
+- (void)appClient:(ARDAppClient *)client
+         didError:(NSError *)error {
+  NSString *message =
+      [NSString stringWithFormat:@"%@", error.localizedDescription];
+  [self hangup];
+  [self showAlertWithMessage:message];
+}
+
+#pragma mark - ARDVideoCallViewDelegate
+
+- (void)videoCallViewDidHangup:(ARDVideoCallView *)view {
+  [self hangup];
+}
+
+- (void)videoCallView:(ARDVideoCallView *)view
+    shouldSwitchCameraWithCompletion:(void (^)(NSError *))completion {
+  [_captureController switchCamera:completion];
+}
+
+- (void)videoCallView:(ARDVideoCallView *)view
+    shouldChangeRouteWithCompletion:(void (^)(void))completion {
+  NSParameterAssert(completion);
+  AVAudioSessionPortOverride override = AVAudioSessionPortOverrideNone;
+  if (_portOverride == AVAudioSessionPortOverrideNone) {
+    override = AVAudioSessionPortOverrideSpeaker;
+  }
+  [RTC_OBJC_TYPE(RTCDispatcher) dispatchAsyncOnType:RTCDispatcherTypeAudioSession
+                                              block:^{
+                                                RTC_OBJC_TYPE(RTCAudioSession) *session =
+                                                    [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+                                                [session lockForConfiguration];
+                                                NSError *error = nil;
+                                                if ([session overrideOutputAudioPort:override
+                                                                               error:&error]) {
+                                                  self.portOverride = override;
+                                                } else {
+                                                  RTCLogError(@"Error overriding output port: %@",
+                                                              error.localizedDescription);
+                                                }
+                                                [session unlockForConfiguration];
+                                                completion();
+                                              }];
+}
+
+- (void)videoCallViewDidEnableStats:(ARDVideoCallView *)view {
+  _client.shouldGetStats = YES;
+  _videoCallView.statsView.hidden = NO;
+}
+
+#pragma mark - RTC_OBJC_TYPE(RTCAudioSessionDelegate)
+
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches {
+  RTCLog(@"Audio session detected glitch, total: %lld", totalNumberOfGlitches);
+}
+
+#pragma mark - Private
+
+- (void)setRemoteVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)remoteVideoTrack {
+  if (_remoteVideoTrack == remoteVideoTrack) {
+    return;
+  }
+  [_remoteVideoTrack removeRenderer:_videoCallView.remoteVideoView];
+  _remoteVideoTrack = nil;
+  [_videoCallView.remoteVideoView renderFrame:nil];
+  _remoteVideoTrack = remoteVideoTrack;
+  [_remoteVideoTrack addRenderer:_videoCallView.remoteVideoView];
+}
+
+- (void)hangup {
+  self.remoteVideoTrack = nil;
+  _videoCallView.localVideoView.captureSession = nil;
+  [_captureController stopCapture];
+  _captureController = nil;
+  [_fileCaptureController stopCapture];
+  _fileCaptureController = nil;
+  [_client disconnect];
+  [_delegate viewControllerDidFinish:self];
+}
+
+- (NSString *)statusTextForState:(RTCIceConnectionState)state {
+  switch (state) {
+    case RTCIceConnectionStateNew:
+    case RTCIceConnectionStateChecking:
+      return @"Connecting...";
+    case RTCIceConnectionStateConnected:
+    case RTCIceConnectionStateCompleted:
+    case RTCIceConnectionStateFailed:
+    case RTCIceConnectionStateDisconnected:
+    case RTCIceConnectionStateClosed:
+    case RTCIceConnectionStateCount:
+      return nil;
+  }
+}
+
+- (void)showAlertWithMessage:(NSString*)message {
+  UIAlertController *alert =
+      [UIAlertController alertControllerWithTitle:nil
+                                          message:message
+                                   preferredStyle:UIAlertControllerStyleAlert];
+
+  UIAlertAction *defaultAction = [UIAlertAction actionWithTitle:@"OK"
+                                                          style:UIAlertActionStyleDefault
+                                                        handler:^(UIAlertAction *action){
+                                                        }];
+
+  [alert addAction:defaultAction];
+  [self presentViewController:alert animated:YES completion:nil];
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/Info.plist b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/Info.plist
new file mode 100644
index 0000000000..a2f0a683ed
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/Info.plist
@@ -0,0 +1,109 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+  <key>BuildMachineOSBuild</key>
+  <string>12E55</string>
+  <key>CFBundleDevelopmentRegion</key>
+  <string>en</string>
+  <key>CFBundleDisplayName</key>
+  <string>AppRTCMobile</string>
+  <key>CFBundleExecutable</key>
+  <string>${EXECUTABLE_NAME}</string>
+  <key>CFBundleIcons</key>
+  <dict>
+    <key>CFBundlePrimaryIcon</key>
+    <dict>
+      <key>CFBundleIconFiles</key>
+      <array>
+        <string>Icon.png</string>
+        <string>Icon-120.png</string>
+        <string>Icon-180.png</string>
+      </array>
+    </dict>
+  </dict>
+  <key>CFBundleIdentifier</key>
+  <string>com.google.AppRTCMobile</string>
+  <key>CFBundleInfoDictionaryVersion</key>
+  <string>6.0</string>
+  <key>CFBundleName</key>
+  <string>${EXECUTABLE_NAME}</string>
+  <key>CFBundlePackageType</key>
+  <string>APPL</string>
+  <key>CFBundleShortVersionString</key>
+  <string>1.0</string>
+  <key>CFBundleSignature</key>
+  <string>????</string>
+  <key>CFBundleSupportedPlatforms</key>
+  <array>
+    <string>iPhoneOS</string>
+  </array>
+  <key>CFBundleVersion</key>
+  <string>1.0</string>
+  <key>UIStatusBarTintParameters</key>
+  <dict>
+    <key>UINavigationBar</key>
+    <dict>
+      <key>Style</key>
+      <string>UIBarStyleDefault</string>
+      <key>Translucent</key>
+      <false/>
+    </dict>
+  </dict>
+  <key>UISupportedInterfaceOrientations</key>
+  <array>
+    <string>UIInterfaceOrientationPortrait</string>
+    <string>UIInterfaceOrientationLandscapeLeft</string>
+    <string>UIInterfaceOrientationLandscapeRight</string>
+    <string>UIInterfaceOrientationPortraitUpsideDown</string>
+  </array>
+  <key>UIAppFonts</key>
+  <array>
+    <string>Roboto-Regular.ttf</string>
+  </array>
+  <key>UIBackgroundModes</key>
+  <array>
+    <string>audio</string>
+    <string>voip</string>
+  </array>
+  <key>NSCameraUsageDescription</key>
+  <string>Camera access needed for video calling</string>
+  <key>NSMicrophoneUsageDescription</key>
+  <string>Microphone access needed for video calling</string>
+  <key>UIFileSharingEnabled</key>
+  <true/>
+  <key>UILaunchImages</key>
+  <array>
+    <dict>
+      <key>UILaunchImageMinimumOSVersion</key>
+      <string>7.0</string>
+      <key>UILaunchImageName</key>
+      <string>iPhone5</string>
+      <key>UILaunchImageOrientation</key>
+      <string>Portrait</string>
+      <key>UILaunchImageSize</key>
+      <string>{320, 568}</string>
+    </dict>
+    <dict>
+      <key>UILaunchImageMinimumOSVersion</key>
+      <string>8.0</string>
+      <key>UILaunchImageName</key>
+      <string>iPhone6</string>
+      <key>UILaunchImageOrientation</key>
+      <string>Portrait</string>
+      <key>UILaunchImageSize</key>
+      <string>{375, 667}</string>
+    </dict>
+    <dict>
+      <key>UILaunchImageMinimumOSVersion</key>
+      <string>8.0</string>
+      <key>UILaunchImageName</key>
+      <string>iPhone6p</string>
+      <key>UILaunchImageOrientation</key>
+      <string>Portrait</string>
+      <key>UILaunchImageSize</key>
+      <string>{414, 736}</string>
+    </dict>
+  </array>
+</dict>
+</plist>
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.h
new file mode 100644
index 0000000000..3a93c253b2
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "sdk/objc/base/RTCVideoCodecInfo.h"
+
+@interface RTC_OBJC_TYPE (RTCVideoCodecInfo)
+(HumanReadable)
+
+    - (NSString *)humanReadableDescription;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.m
new file mode 100644
index 0000000000..5e0c52c5c4
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/RTCVideoCodecInfo+HumanReadable.m
@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoCodecInfo+HumanReadable.h"
+
+#import "sdk/objc/components/video_codec/RTCH264ProfileLevelId.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoCodecInfo)
+(HumanReadable)
+
+    - (NSString *)humanReadableDescription {
+  if ([self.name isEqualToString:@"H264"]) {
+    NSString *profileId = self.parameters[@"profile-level-id"];
+    RTC_OBJC_TYPE(RTCH264ProfileLevelId) *profileLevelId =
+        [[RTC_OBJC_TYPE(RTCH264ProfileLevelId) alloc] initWithHexString:profileId];
+    if (profileLevelId.profile == RTCH264ProfileConstrainedHigh ||
+        profileLevelId.profile == RTCH264ProfileHigh) {
+      return @"H264 (High)";
+    } else if (profileLevelId.profile == RTCH264ProfileConstrainedBaseline ||
+               profileLevelId.profile == RTCH264ProfileBaseline) {
+      return @"H264 (Baseline)";
+    } else {
+      return [NSString stringWithFormat:@"H264 (%@)", profileId];
+    }
+  } else {
+    return self.name;
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.h
new file mode 100644
index 0000000000..d56ba02c2e
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+@interface UIImage (ARDUtilities)
+
+// Returns an color tinted version for the given image resource.
++ (UIImage *)imageForName:(NSString *)name color:(UIColor *)color;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.m
new file mode 100644
index 0000000000..1bbe8c342f
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.m
@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "UIImage+ARDUtilities.h"
+
+@implementation UIImage (ARDUtilities)
+
++ (UIImage *)imageForName:(NSString *)name color:(UIColor *)color {
+  UIImage *image = [UIImage imageNamed:name];
+  if (!image) {
+    return nil;
+  }
+  UIGraphicsBeginImageContextWithOptions(image.size, NO, 0.0f);
+  [color setFill];
+  CGRect bounds = CGRectMake(0, 0, image.size.width, image.size.height);
+  UIRectFill(bounds);
+  [image drawInRect:bounds blendMode:kCGBlendModeDestinationIn alpha:1.0f];
+  UIImage *coloredImage = UIGraphicsGetImageFromCurrentImageContext();
+  UIGraphicsEndImageContext();
+
+  return coloredImage;
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h
new file mode 100644
index 0000000000..2c4a56368a
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <ReplayKit/ReplayKit.h>
+
+#import "sdk/objc/base/RTCLogging.h"
+
+#import "ARDAppClient.h"
+
+@protocol ARDExternalSampleDelegate;
+
+API_AVAILABLE(ios(10.0))
+@interface ARDBroadcastSampleHandler : RPBroadcastSampleHandler <ARDAppClientDelegate>
+
+@property(nonatomic, strong) id<ARDExternalSampleDelegate> capturer;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.m
new file mode 100644
index 0000000000..1c276d965f
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.m
@@ -0,0 +1,130 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDBroadcastSampleHandler.h"
+
+#import <os/log.h>
+
+#import "ARDExternalSampleCapturer.h"
+#import "ARDSettingsModel.h"
+
+#import "sdk/objc/api/logging/RTCCallbackLogger.h"
+#import "sdk/objc/base/RTCLogging.h"
+
+@implementation ARDBroadcastSampleHandler {
+  ARDAppClient *_client;
+  RTC_OBJC_TYPE(RTCCallbackLogger) * _callbackLogger;
+}
+
+@synthesize capturer = _capturer;
+
+- (instancetype)init {
+  if (self = [super init]) {
+    _callbackLogger = [[RTC_OBJC_TYPE(RTCCallbackLogger) alloc] init];
+    os_log_t rtc_os_log = os_log_create("com.google.AppRTCMobile", "RTCLog");
+    [_callbackLogger start:^(NSString *logMessage) {
+      os_log(rtc_os_log, "%{public}s", [logMessage cStringUsingEncoding:NSUTF8StringEncoding]);
+    }];
+  }
+  return self;
+}
+
+- (void)broadcastStartedWithSetupInfo:(NSDictionary<NSString *, NSObject *> *)setupInfo {
+  // User has requested to start the broadcast. Setup info from the UI extension can be supplied but
+  // optional.
+  ARDSettingsModel *settingsModel = [[ARDSettingsModel alloc] init];
+
+  _client = [[ARDAppClient alloc] initWithDelegate:self];
+  _client.broadcast = YES;
+
+  NSString *roomName = nil;
+  if (setupInfo[@"roomName"]) {
+    roomName = (NSString *)setupInfo[@"roomName"];
+  } else {
+    u_int32_t randomRoomSuffix = arc4random_uniform(1000);
+    roomName = [NSString stringWithFormat:@"broadcast_%d", randomRoomSuffix];
+  }
+  [_client connectToRoomWithId:roomName settings:settingsModel isLoopback:NO];
+  RTCLog(@"Broadcast started.");
+}
+
+- (void)broadcastPaused {
+  // User has requested to pause the broadcast. Samples will stop being delivered.
+}
+
+- (void)broadcastResumed {
+  // User has requested to resume the broadcast. Samples delivery will resume.
+}
+
+- (void)broadcastFinished {
+  // User has requested to finish the broadcast.
+  [_client disconnect];
+}
+
+- (void)processSampleBuffer:(CMSampleBufferRef)sampleBuffer
+                   withType:(RPSampleBufferType)sampleBufferType {
+  switch (sampleBufferType) {
+    case RPSampleBufferTypeVideo:
+      [self.capturer didCaptureSampleBuffer:sampleBuffer];
+      break;
+    case RPSampleBufferTypeAudioApp:
+      break;
+    case RPSampleBufferTypeAudioMic:
+      break;
+    default:
+      break;
+  }
+}
+
+#pragma mark - ARDAppClientDelegate
+
+- (void)appClient:(ARDAppClient *)client didChangeState:(ARDAppClientState)state {
+  switch (state) {
+    case kARDAppClientStateConnected:
+      RTCLog(@"Client connected.");
+      break;
+    case kARDAppClientStateConnecting:
+      RTCLog("Client connecting.");
+      break;
+    case kARDAppClientStateDisconnected:
+      RTCLog(@"Client disconnected.");
+      break;
+  }
+}
+
+- (void)appClient:(ARDAppClient *)client didChangeConnectionState:(RTCIceConnectionState)state {
+  RTCLog(@"ICE state changed: %ld", (long)state);
+}
+
+- (void)appClient:(ARDAppClient *)client
+    didCreateLocalCapturer:(RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)localCapturer {
+}
+
+- (void)appClient:(ARDAppClient *)client
+    didCreateLocalExternalSampleCapturer:(ARDExternalSampleCapturer *)externalSampleCapturer {
+  self.capturer = externalSampleCapturer;
+}
+
+- (void)appClient:(ARDAppClient *)client
+    didReceiveLocalVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)localVideoTrack {
+}
+
+- (void)appClient:(ARDAppClient *)client
+    didReceiveRemoteVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)remoteVideoTrack {
+}
+
+- (void)appClient:(ARDAppClient *)client didGetStats:(RTC_OBJC_TYPE(RTCStatisticsReport) *)stats {
+}
+
+- (void)appClient:(ARDAppClient *)client didError:(NSError *)error {
+  RTCLog(@"Error: %@", error);
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.h
new file mode 100644
index 0000000000..bbf397d8a9
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <ReplayKit/ReplayKit.h>
+#import <UIKit/UIKit.h>
+
+API_AVAILABLE(ios(11.0))
+@interface ARDBroadcastSetupViewController : UIViewController <UITextFieldDelegate>
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.m
new file mode 100644
index 0000000000..55438f17d8
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.m
@@ -0,0 +1,107 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ARDBroadcastSetupViewController.h"
+
+@implementation ARDBroadcastSetupViewController {
+  UITextField *_roomNameField;
+}
+
+- (void)loadView {
+  UIView *view = [[UIView alloc] initWithFrame:CGRectZero];
+  view.backgroundColor = [UIColor colorWithWhite:1.0 alpha:0.7];
+
+  UIImageView *imageView = [[UIImageView alloc] initWithImage:[UIImage imageNamed:@"Icon-180"]];
+  imageView.translatesAutoresizingMaskIntoConstraints = NO;
+  [view addSubview:imageView];
+
+  _roomNameField = [[UITextField alloc] initWithFrame:CGRectZero];
+  _roomNameField.borderStyle = UITextBorderStyleRoundedRect;
+  _roomNameField.font = [UIFont systemFontOfSize:14.0];
+  _roomNameField.translatesAutoresizingMaskIntoConstraints = NO;
+  _roomNameField.placeholder = @"Room name";
+  _roomNameField.returnKeyType = UIReturnKeyDone;
+  _roomNameField.delegate = self;
+  [view addSubview:_roomNameField];
+
+  UIButton *doneButton = [UIButton buttonWithType:UIButtonTypeSystem];
+  doneButton.translatesAutoresizingMaskIntoConstraints = NO;
+  doneButton.titleLabel.font = [UIFont systemFontOfSize:20.0];
+  [doneButton setTitle:@"Done" forState:UIControlStateNormal];
+  [doneButton addTarget:self
+                 action:@selector(userDidFinishSetup)
+       forControlEvents:UIControlEventTouchUpInside];
+  [view addSubview:doneButton];
+
+  UIButton *cancelButton = [UIButton buttonWithType:UIButtonTypeSystem];
+  cancelButton.translatesAutoresizingMaskIntoConstraints = NO;
+  cancelButton.titleLabel.font = [UIFont systemFontOfSize:20.0];
+  [cancelButton setTitle:@"Cancel" forState:UIControlStateNormal];
+  [cancelButton addTarget:self
+                   action:@selector(userDidCancelSetup)
+         forControlEvents:UIControlEventTouchUpInside];
+  [view addSubview:cancelButton];
+
+  UILayoutGuide *margin = view.layoutMarginsGuide;
+  [imageView.widthAnchor constraintEqualToConstant:60.0].active = YES;
+  [imageView.heightAnchor constraintEqualToConstant:60.0].active = YES;
+  [imageView.topAnchor constraintEqualToAnchor:margin.topAnchor constant:20].active = YES;
+  [imageView.centerXAnchor constraintEqualToAnchor:view.centerXAnchor].active = YES;
+
+  [_roomNameField.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor].active = YES;
+  [_roomNameField.topAnchor constraintEqualToAnchor:imageView.bottomAnchor constant:20].active =
+      YES;
+  [_roomNameField.trailingAnchor constraintEqualToAnchor:margin.trailingAnchor].active = YES;
+
+  [doneButton.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor].active = YES;
+  [doneButton.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor constant:-20].active = YES;
+
+  [cancelButton.trailingAnchor constraintEqualToAnchor:margin.trailingAnchor].active = YES;
+  [cancelButton.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor constant:-20].active = YES;
+
+  UITapGestureRecognizer *tgr =
+      [[UITapGestureRecognizer alloc] initWithTarget:self action:@selector(didTap:)];
+  [view addGestureRecognizer:tgr];
+
+  self.view = view;
+}
+
+- (IBAction)didTap:(id)sender {
+  [self.view endEditing:YES];
+}
+
+- (void)userDidFinishSetup {
+  // URL of the resource where broadcast can be viewed that will be returned to the application
+  NSURL *broadcastURL = [NSURL
+      URLWithString:[NSString stringWithFormat:@"https://appr.tc/r/%@", _roomNameField.text]];
+
+  // Dictionary with setup information that will be provided to broadcast extension when broadcast
+  // is started
+  NSDictionary *setupInfo = @{@"roomName" : _roomNameField.text};
+
+  // Tell ReplayKit that the extension is finished setting up and can begin broadcasting
+  [self.extensionContext completeRequestWithBroadcastURL:broadcastURL setupInfo:setupInfo];
+}
+
+- (void)userDidCancelSetup {
+  // Tell ReplayKit that the extension was cancelled by the user
+  [self.extensionContext cancelRequestWithError:[NSError errorWithDomain:@"com.google.AppRTCMobile"
+                                                                    code:-1
+                                                                userInfo:nil]];
+}
+
+#pragma mark - UITextFieldDelegate
+
+- (BOOL)textFieldShouldReturn:(UITextField *)textField {
+  [self userDidFinishSetup];
+  return YES;
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastSetupUIInfo.plist b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastSetupUIInfo.plist
new file mode 100644
index 0000000000..a123c111e5
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastSetupUIInfo.plist
@@ -0,0 +1,39 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>CFBundleDevelopmentRegion</key>
+	<string>en</string>
+	<key>CFBundleDisplayName</key>
+	<string>AppRTCMobile</string>
+	<key>CFBundleExecutable</key>
+	<string>$(EXECUTABLE_NAME)</string>
+	<key>CFBundleIdentifier</key>
+	<string>com.google.AppRTCMobile.BroadcastSetupUI</string>
+	<key>CFBundleInfoDictionaryVersion</key>
+	<string>6.0</string>
+	<key>CFBundleName</key>
+	<string>$(PRODUCT_NAME)</string>
+	<key>CFBundlePackageType</key>
+	<string>XPC!</string>
+	<key>CFBundleShortVersionString</key>
+	<string>1.0</string>
+	<key>CFBundleVersion</key>
+	<string>1</string>
+	<key>NSExtension</key>
+	<dict>
+		<key>NSExtensionAttributes</key>
+		<dict>
+			<key>NSExtensionActivationRule</key>
+			<dict>
+				<key>NSExtensionActivationSupportsReplayKitStreaming</key>
+				<true/>
+			</dict>
+		</dict>
+		<key>NSExtensionPointIdentifier</key>
+		<string>com.apple.broadcast-services-setupui</string>
+		<key>NSExtensionPrincipalClass</key>
+		<string>ARDBroadcastSetupViewController</string>
+	</dict>
+</dict>
+</plist>
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastUploadInfo.plist b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastUploadInfo.plist
new file mode 100644
index 0000000000..2bab60ea8f
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastUploadInfo.plist
@@ -0,0 +1,33 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>CFBundleDevelopmentRegion</key>
+	<string>en</string>
+	<key>CFBundleDisplayName</key>
+	<string>AppRTCMobile</string>
+	<key>CFBundleExecutable</key>
+	<string>$(EXECUTABLE_NAME)</string>
+	<key>CFBundleIdentifier</key>
+	<string>com.google.AppRTCMobile.BroadcastUpload</string>
+	<key>CFBundleInfoDictionaryVersion</key>
+	<string>6.0</string>
+	<key>CFBundleName</key>
+	<string>$(PRODUCT_NAME)</string>
+	<key>CFBundlePackageType</key>
+	<string>XPC!</string>
+	<key>CFBundleShortVersionString</key>
+	<string>1.0</string>
+	<key>CFBundleVersion</key>
+	<string>1</string>
+	<key>NSExtension</key>
+	<dict>
+		<key>NSExtensionPointIdentifier</key>
+		<string>com.apple.broadcast-services-upload</string>
+		<key>NSExtensionPrincipalClass</key>
+		<string>ARDBroadcastSampleHandler</string>
+		<key>RPBroadcastProcessMode</key>
+		<string>RPBroadcastProcessModeSampleBuffer</string>
+	</dict>
+</dict>
+</plist>
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/main.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/main.m
new file mode 100644
index 0000000000..00b83f7fd2
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/main.m
@@ -0,0 +1,20 @@
+/*
+ *  Copyright 2013 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+#import "ARDAppDelegate.h"
+
+int main(int argc, char* argv[]) {
+  @autoreleasepool {
+    return UIApplicationMain(
+        argc, argv, nil, NSStringFromClass([ARDAppDelegate class]));
+  }
+}
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/Roboto-Regular.ttf b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/Roboto-Regular.ttf
new file mode 100644
index 0000000000..0e58508a64
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/Roboto-Regular.ttf
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/foreman.mp4 b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/foreman.mp4
new file mode 100644
index 0000000000..ccffbf4722
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/foreman.mp4
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/iPhone5@2x.png b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/iPhone5@2x.png
new file mode 100644
index 0000000000..9d005fde06
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/iPhone5@2x.png
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/iPhone6@2x.png b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/iPhone6@2x.png
new file mode 100644
index 0000000000..fce3eb95b3
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/iPhone6@2x.png
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/iPhone6p@3x.png b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/iPhone6p@3x.png
new file mode 100644
index 0000000000..aee20c2209
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/iPhone6p@3x.png
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_call_end_black_24dp.png b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_call_end_black_24dp.png
new file mode 100644
index 0000000000..531cb0f280
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_call_end_black_24dp.png
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_call_end_black_24dp@2x.png b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_call_end_black_24dp@2x.png
new file mode 100644
index 0000000000..03dd381c10
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_call_end_black_24dp@2x.png
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_clear_black_24dp.png b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_clear_black_24dp.png
new file mode 100644
index 0000000000..4ebf8a2270
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_clear_black_24dp.png
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_clear_black_24dp@2x.png b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_clear_black_24dp@2x.png
new file mode 100644
index 0000000000..ed2b2525fd
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_clear_black_24dp@2x.png
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_settings_black_24dp.png b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_settings_black_24dp.png
new file mode 100644
index 0000000000..c59419c02b
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_settings_black_24dp.png
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_settings_black_24dp@2x.png b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_settings_black_24dp@2x.png
new file mode 100644
index 0000000000..e84e188a1d
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_settings_black_24dp@2x.png
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_surround_sound_black_24dp.png b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_surround_sound_black_24dp.png
new file mode 100644
index 0000000000..8f3343d3a7
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_surround_sound_black_24dp.png
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_surround_sound_black_24dp@2x.png b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_surround_sound_black_24dp@2x.png
new file mode 100644
index 0000000000..764880467a
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_surround_sound_black_24dp@2x.png
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_switch_video_black_24dp.png b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_switch_video_black_24dp.png
new file mode 100644
index 0000000000..85271c8253
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_switch_video_black_24dp.png
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_switch_video_black_24dp@2x.png b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_switch_video_black_24dp@2x.png
new file mode 100644
index 0000000000..62b13a6a09
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/ic_switch_video_black_24dp@2x.png
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/mozart.mp3 b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/mozart.mp3
new file mode 100644
index 0000000000..5981ba3a91
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/ios/resources/mozart.mp3
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.h
new file mode 100644
index 0000000000..2b3ce094a2
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.h
@@ -0,0 +1,14 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Cocoa/Cocoa.h>
+
+@interface APPRTCAppDelegate : NSObject <NSApplicationDelegate>
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.m
new file mode 100644
index 0000000000..36a470021d
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.m
@@ -0,0 +1,55 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "APPRTCAppDelegate.h"
+#import "APPRTCViewController.h"
+#import "sdk/objc/api/peerconnection/RTCSSLAdapter.h"
+
+@interface APPRTCAppDelegate () <NSWindowDelegate>
+@end
+
+@implementation APPRTCAppDelegate {
+  APPRTCViewController* _viewController;
+  NSWindow* _window;
+}
+
+#pragma mark - NSApplicationDelegate
+
+- (void)applicationDidFinishLaunching:(NSNotification*)notification {
+  RTCInitializeSSL();
+  NSScreen* screen = [NSScreen mainScreen];
+  NSRect visibleRect = [screen visibleFrame];
+  NSRect windowRect = NSMakeRect(NSMidX(visibleRect),
+                                 NSMidY(visibleRect),
+                                 1320,
+                                 1140);
+  NSUInteger styleMask = NSWindowStyleMaskTitled | NSWindowStyleMaskClosable;
+  _window = [[NSWindow alloc] initWithContentRect:windowRect
+                                        styleMask:styleMask
+                                          backing:NSBackingStoreBuffered
+                                            defer:NO];
+  _window.delegate = self;
+  [_window makeKeyAndOrderFront:self];
+  [_window makeMainWindow];
+  _viewController = [[APPRTCViewController alloc] initWithNibName:nil
+                                                           bundle:nil];
+  [_window setContentView:[_viewController view]];
+}
+
+#pragma mark - NSWindow
+
+- (void)windowWillClose:(NSNotification*)notification {
+  [_viewController windowWillClose:notification];
+  RTCCleanupSSL();
+  [NSApp terminate:self];
+}
+
+@end
+
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.h
new file mode 100644
index 0000000000..306ecd9c7f
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AppKit/AppKit.h>
+
+@interface APPRTCViewController : NSViewController
+
+- (void)windowWillClose:(NSNotification*)notification;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.m
new file mode 100644
index 0000000000..8904187215
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.m
@@ -0,0 +1,438 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "APPRTCViewController.h"
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "sdk/objc/api/peerconnection/RTCVideoTrack.h"
+#import "sdk/objc/components/renderer/metal/RTCMTLNSVideoView.h"
+#import "sdk/objc/components/renderer/opengl/RTCNSGLVideoView.h"
+
+#import "ARDAppClient.h"
+#import "ARDCaptureController.h"
+#import "ARDSettingsModel.h"
+
+static NSUInteger const kContentWidth = 900;
+static NSUInteger const kRoomFieldWidth = 200;
+static NSUInteger const kActionItemHeight = 30;
+static NSUInteger const kBottomViewHeight = 200;
+
+@class APPRTCMainView;
+@protocol APPRTCMainViewDelegate
+
+- (void)appRTCMainView:(APPRTCMainView*)mainView
+        didEnterRoomId:(NSString*)roomId
+              loopback:(BOOL)isLoopback;
+
+@end
+
+@interface APPRTCMainView : NSView
+
+@property(nonatomic, weak) id<APPRTCMainViewDelegate> delegate;
+@property(nonatomic, readonly) NSView<RTC_OBJC_TYPE(RTCVideoRenderer)>* localVideoView;
+@property(nonatomic, readonly) NSView<RTC_OBJC_TYPE(RTCVideoRenderer)>* remoteVideoView;
+@property(nonatomic, readonly) NSTextView* logView;
+
+- (void)displayLogMessage:(NSString*)message;
+
+@end
+
+@interface APPRTCMainView () <NSTextFieldDelegate, RTC_OBJC_TYPE (RTCNSGLVideoViewDelegate)>
+@end
+@implementation APPRTCMainView  {
+  NSScrollView* _scrollView;
+  NSView* _actionItemsView;
+  NSButton* _connectButton;
+  NSButton* _loopbackButton;
+  NSTextField* _roomField;
+  CGSize _localVideoSize;
+  CGSize _remoteVideoSize;
+}
+
+@synthesize delegate = _delegate;
+@synthesize localVideoView = _localVideoView;
+@synthesize remoteVideoView = _remoteVideoView;
+@synthesize logView = _logView;
+
+- (void)displayLogMessage:(NSString *)message {
+  dispatch_async(dispatch_get_main_queue(), ^{
+    self.logView.string = [NSString stringWithFormat:@"%@%@\n", self.logView.string, message];
+    NSRange range = NSMakeRange(self.logView.string.length, 0);
+    [self.logView scrollRangeToVisible:range];
+  });
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithFrame:(NSRect)frame {
+  if (self = [super initWithFrame:frame]) {
+    [self setupViews];
+  }
+  return self;
+}
+
++ (BOOL)requiresConstraintBasedLayout {
+  return YES;
+}
+
+- (void)updateConstraints {
+  NSParameterAssert(
+      _roomField != nil &&
+      _scrollView != nil &&
+      _remoteVideoView != nil &&
+      _localVideoView != nil &&
+      _actionItemsView!= nil &&
+      _connectButton != nil &&
+      _loopbackButton != nil);
+
+  [self removeConstraints:[self constraints]];
+  NSDictionary* viewsDictionary =
+      NSDictionaryOfVariableBindings(_roomField,
+                                     _scrollView,
+                                     _remoteVideoView,
+                                     _localVideoView,
+                                     _actionItemsView,
+                                     _connectButton,
+                                     _loopbackButton);
+
+  NSSize remoteViewSize = [self remoteVideoViewSize];
+  NSDictionary* metrics = @{
+    @"remoteViewWidth" : @(remoteViewSize.width),
+    @"remoteViewHeight" : @(remoteViewSize.height),
+    @"kBottomViewHeight" : @(kBottomViewHeight),
+    @"localViewHeight" : @(remoteViewSize.height / 3),
+    @"localViewWidth" : @(remoteViewSize.width / 3),
+    @"kRoomFieldWidth" : @(kRoomFieldWidth),
+    @"kActionItemHeight" : @(kActionItemHeight)
+  };
+  // Declare this separately to avoid compiler warning about splitting string
+  // within an NSArray expression.
+  NSString* verticalConstraintLeft =
+      @"V:|-[_remoteVideoView(remoteViewHeight)]-[_scrollView(kBottomViewHeight)]-|";
+  NSString* verticalConstraintRight =
+      @"V:|-[_remoteVideoView(remoteViewHeight)]-[_actionItemsView(kBottomViewHeight)]-|";
+  NSArray* constraintFormats = @[
+      verticalConstraintLeft,
+      verticalConstraintRight,
+      @"H:|-[_remoteVideoView(remoteViewWidth)]-|",
+      @"V:|-[_localVideoView(localViewHeight)]",
+      @"H:|-[_localVideoView(localViewWidth)]",
+      @"H:|-[_scrollView(==_actionItemsView)]-[_actionItemsView]-|"
+  ];
+
+  NSArray* actionItemsConstraints = @[
+      @"H:|-[_roomField(kRoomFieldWidth)]-[_loopbackButton(kRoomFieldWidth)]",
+      @"H:|-[_connectButton(kRoomFieldWidth)]",
+      @"V:|-[_roomField(kActionItemHeight)]-[_connectButton(kActionItemHeight)]",
+      @"V:|-[_loopbackButton(kActionItemHeight)]",
+      ];
+
+  [APPRTCMainView addConstraints:constraintFormats
+                          toView:self
+                 viewsDictionary:viewsDictionary
+                         metrics:metrics];
+  [APPRTCMainView addConstraints:actionItemsConstraints
+                          toView:_actionItemsView
+                 viewsDictionary:viewsDictionary
+                         metrics:metrics];
+  [super updateConstraints];
+}
+
+#pragma mark - Constraints helper
+
++ (void)addConstraints:(NSArray*)constraints toView:(NSView*)view
+       viewsDictionary:(NSDictionary*)viewsDictionary
+               metrics:(NSDictionary*)metrics {
+  for (NSString* constraintFormat in constraints) {
+    NSArray* constraints =
+    [NSLayoutConstraint constraintsWithVisualFormat:constraintFormat
+                                            options:0
+                                            metrics:metrics
+                                              views:viewsDictionary];
+    for (NSLayoutConstraint* constraint in constraints) {
+      [view addConstraint:constraint];
+    }
+  }
+}
+
+#pragma mark - Control actions
+
+- (void)startCall:(id)sender {
+  NSString* roomString = _roomField.stringValue;
+  // Generate room id for loopback options.
+  if (_loopbackButton.intValue && [roomString isEqualToString:@""]) {
+    roomString = [NSUUID UUID].UUIDString;
+    roomString = [roomString stringByReplacingOccurrencesOfString:@"-" withString:@""];
+  }
+  [self.delegate appRTCMainView:self
+                 didEnterRoomId:roomString
+                       loopback:_loopbackButton.intValue];
+  [self setNeedsUpdateConstraints:YES];
+}
+
+#pragma mark - RTC_OBJC_TYPE(RTCNSGLVideoViewDelegate)
+
+- (void)videoView:(RTC_OBJC_TYPE(RTCNSGLVideoView) *)videoView didChangeVideoSize:(NSSize)size {
+  if (videoView == _remoteVideoView) {
+    _remoteVideoSize = size;
+  } else if (videoView == _localVideoView) {
+    _localVideoSize = size;
+  } else {
+    return;
+  }
+
+  [self setNeedsUpdateConstraints:YES];
+}
+
+#pragma mark - Private
+
+- (void)setupViews {
+  NSParameterAssert([[self subviews] count] == 0);
+
+  _logView = [[NSTextView alloc] initWithFrame:NSZeroRect];
+  [_logView setMinSize:NSMakeSize(0, kBottomViewHeight)];
+  [_logView setMaxSize:NSMakeSize(FLT_MAX, FLT_MAX)];
+  [_logView setVerticallyResizable:YES];
+  [_logView setAutoresizingMask:NSViewWidthSizable];
+  NSTextContainer* textContainer = [_logView textContainer];
+  NSSize containerSize = NSMakeSize(kContentWidth, FLT_MAX);
+  [textContainer setContainerSize:containerSize];
+  [textContainer setWidthTracksTextView:YES];
+  [_logView setEditable:NO];
+
+  [self setupActionItemsView];
+
+  _scrollView = [[NSScrollView alloc] initWithFrame:NSZeroRect];
+  [_scrollView setTranslatesAutoresizingMaskIntoConstraints:NO];
+  [_scrollView setHasVerticalScroller:YES];
+  [_scrollView setDocumentView:_logView];
+  [self addSubview:_scrollView];
+
+// NOTE (daniela): Ignoring Clang diagonstic here.
+// We're performing run time check to make sure class is available on runtime.
+// If not we're providing sensible default.
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wpartial-availability"
+  if ([RTC_OBJC_TYPE(RTCMTLNSVideoView) class] &&
+      [RTC_OBJC_TYPE(RTCMTLNSVideoView) isMetalAvailable]) {
+    _remoteVideoView = [[RTC_OBJC_TYPE(RTCMTLNSVideoView) alloc] initWithFrame:NSZeroRect];
+    _localVideoView = [[RTC_OBJC_TYPE(RTCMTLNSVideoView) alloc] initWithFrame:NSZeroRect];
+  }
+#pragma clang diagnostic pop
+  if (_remoteVideoView == nil) {
+    NSOpenGLPixelFormatAttribute attributes[] = {
+      NSOpenGLPFADoubleBuffer,
+      NSOpenGLPFADepthSize, 24,
+      NSOpenGLPFAOpenGLProfile,
+      NSOpenGLProfileVersion3_2Core,
+      0
+    };
+    NSOpenGLPixelFormat* pixelFormat =
+    [[NSOpenGLPixelFormat alloc] initWithAttributes:attributes];
+
+    RTC_OBJC_TYPE(RTCNSGLVideoView)* remote =
+        [[RTC_OBJC_TYPE(RTCNSGLVideoView) alloc] initWithFrame:NSZeroRect pixelFormat:pixelFormat];
+    remote.delegate = self;
+    _remoteVideoView = remote;
+
+    RTC_OBJC_TYPE(RTCNSGLVideoView)* local =
+        [[RTC_OBJC_TYPE(RTCNSGLVideoView) alloc] initWithFrame:NSZeroRect pixelFormat:pixelFormat];
+    local.delegate = self;
+    _localVideoView = local;
+  }
+
+  [_remoteVideoView setTranslatesAutoresizingMaskIntoConstraints:NO];
+  [self addSubview:_remoteVideoView];
+  [_localVideoView setTranslatesAutoresizingMaskIntoConstraints:NO];
+  [self addSubview:_localVideoView];
+}
+
+- (void)setupActionItemsView {
+  _actionItemsView = [[NSView alloc] initWithFrame:NSZeroRect];
+  [_actionItemsView setTranslatesAutoresizingMaskIntoConstraints:NO];
+  [self addSubview:_actionItemsView];
+
+  _roomField = [[NSTextField alloc] initWithFrame:NSZeroRect];
+  [_roomField setTranslatesAutoresizingMaskIntoConstraints:NO];
+  [[_roomField cell] setPlaceholderString: @"Enter AppRTC room id"];
+  [_actionItemsView addSubview:_roomField];
+  [_roomField setEditable:YES];
+
+  _connectButton = [[NSButton alloc] initWithFrame:NSZeroRect];
+  [_connectButton setTranslatesAutoresizingMaskIntoConstraints:NO];
+  _connectButton.title = @"Start call";
+  _connectButton.bezelStyle = NSRoundedBezelStyle;
+  _connectButton.target = self;
+  _connectButton.action = @selector(startCall:);
+  [_actionItemsView addSubview:_connectButton];
+
+  _loopbackButton = [[NSButton alloc] initWithFrame:NSZeroRect];
+  [_loopbackButton setTranslatesAutoresizingMaskIntoConstraints:NO];
+  _loopbackButton.title = @"Loopback";
+  [_loopbackButton setButtonType:NSSwitchButton];
+  [_actionItemsView addSubview:_loopbackButton];
+}
+
+- (NSSize)remoteVideoViewSize {
+  if (!_remoteVideoView.bounds.size.width) {
+    return NSMakeSize(kContentWidth, 0);
+  }
+  NSInteger width = MAX(_remoteVideoView.bounds.size.width, kContentWidth);
+  NSInteger height = (width/16) * 9;
+  return NSMakeSize(width, height);
+}
+
+@end
+
+@interface APPRTCViewController ()
+    <ARDAppClientDelegate, APPRTCMainViewDelegate>
+@property(nonatomic, readonly) APPRTCMainView* mainView;
+@end
+
+@implementation APPRTCViewController {
+  ARDAppClient* _client;
+  RTC_OBJC_TYPE(RTCVideoTrack) * _localVideoTrack;
+  RTC_OBJC_TYPE(RTCVideoTrack) * _remoteVideoTrack;
+  ARDCaptureController* _captureController;
+}
+
+- (void)dealloc {
+  [self disconnect];
+}
+
+- (void)viewDidAppear {
+  [super viewDidAppear];
+  [self displayUsageInstructions];
+}
+
+- (void)loadView {
+  APPRTCMainView* view = [[APPRTCMainView alloc] initWithFrame:NSZeroRect];
+  [view setTranslatesAutoresizingMaskIntoConstraints:NO];
+  view.delegate = self;
+  self.view = view;
+}
+
+- (void)windowWillClose:(NSNotification*)notification {
+  [self disconnect];
+}
+
+#pragma mark - Usage
+
+- (void)displayUsageInstructions {
+  [self.mainView displayLogMessage:
+   @"To start call:\n"
+   @"• Enter AppRTC room id (not neccessary for loopback)\n"
+   @"• Start call"];
+}
+
+#pragma mark - ARDAppClientDelegate
+
+- (void)appClient:(ARDAppClient *)client
+    didChangeState:(ARDAppClientState)state {
+  switch (state) {
+    case kARDAppClientStateConnected:
+      [self.mainView displayLogMessage:@"Client connected."];
+      break;
+    case kARDAppClientStateConnecting:
+      [self.mainView displayLogMessage:@"Client connecting."];
+      break;
+    case kARDAppClientStateDisconnected:
+      [self.mainView displayLogMessage:@"Client disconnected."];
+      [self resetUI];
+      _client = nil;
+      break;
+  }
+}
+
+- (void)appClient:(ARDAppClient *)client
+    didChangeConnectionState:(RTCIceConnectionState)state {
+}
+
+- (void)appClient:(ARDAppClient*)client
+    didCreateLocalCapturer:(RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)localCapturer {
+  _captureController =
+      [[ARDCaptureController alloc] initWithCapturer:localCapturer
+                                            settings:[[ARDSettingsModel alloc] init]];
+  [_captureController startCapture];
+}
+
+- (void)appClient:(ARDAppClient*)client
+    didReceiveLocalVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)localVideoTrack {
+  _localVideoTrack = localVideoTrack;
+  [_localVideoTrack addRenderer:self.mainView.localVideoView];
+}
+
+- (void)appClient:(ARDAppClient*)client
+    didReceiveRemoteVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)remoteVideoTrack {
+  _remoteVideoTrack = remoteVideoTrack;
+  [_remoteVideoTrack addRenderer:self.mainView.remoteVideoView];
+}
+
+- (void)appClient:(ARDAppClient *)client
+         didError:(NSError *)error {
+  [self showAlertWithMessage:[NSString stringWithFormat:@"%@", error]];
+  [self disconnect];
+}
+
+- (void)appClient:(ARDAppClient *)client
+      didGetStats:(NSArray *)stats {
+}
+
+#pragma mark - APPRTCMainViewDelegate
+
+- (void)appRTCMainView:(APPRTCMainView*)mainView
+        didEnterRoomId:(NSString*)roomId
+              loopback:(BOOL)isLoopback {
+
+  if ([roomId isEqualToString:@""]) {
+    [self.mainView displayLogMessage:@"Missing room id"];
+    return;
+  }
+
+  [self disconnect];
+  ARDAppClient* client = [[ARDAppClient alloc] initWithDelegate:self];
+  [client connectToRoomWithId:roomId
+                     settings:[[ARDSettingsModel alloc] init]  // Use default settings.
+                   isLoopback:isLoopback];
+  _client = client;
+}
+
+#pragma mark - Private
+
+- (APPRTCMainView*)mainView {
+  return (APPRTCMainView*)self.view;
+}
+
+- (void)showAlertWithMessage:(NSString*)message {
+  dispatch_async(dispatch_get_main_queue(), ^{
+    NSAlert* alert = [[NSAlert alloc] init];
+    [alert setMessageText:message];
+    [alert runModal];
+  });
+}
+
+- (void)resetUI {
+  [_remoteVideoTrack removeRenderer:self.mainView.remoteVideoView];
+  [_localVideoTrack removeRenderer:self.mainView.localVideoView];
+  _remoteVideoTrack = nil;
+  _localVideoTrack = nil;
+  [self.mainView.remoteVideoView renderFrame:nil];
+  [self.mainView.localVideoView renderFrame:nil];
+}
+
+- (void)disconnect {
+  [self resetUI];
+  [_captureController stopCapture];
+  _captureController = nil;
+  [_client disconnect];
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/Info.plist b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/Info.plist
new file mode 100644
index 0000000000..d2970eba74
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/Info.plist
@@ -0,0 +1,33 @@
+<?xml version="1.0" encoding="UTF-8"?>
+
+<!DOCTYPE plist PUBLIC "-//Apple/DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+  <key>CFBundleDevelopmentRegion</key>
+  <string>en</string>
+  <key>CFBundleDisplayName</key>
+  <string>${PRODUCT_NAME}</string>
+  <key>CFBundleExecutable</key>
+  <string>${EXECUTABLE_NAME}</string>
+  <key>CFBundleIdentifier</key>
+  <string>com.Google.${PRODUCT_NAME:rfc1034identifier}</string>
+  <key>CFBundleInfoDictionaryVersion</key>
+  <string>6.0</string>
+  <key>CFBundleName</key>
+  <string>${PRODUCT_NAME}</string>
+  <key>CFBundlePackageType</key>
+  <string>APPL</string>
+  <key>CFBundleShortVersionString</key>
+  <string>1.0</string>
+  <key>CFBundleVersion</key>
+  <string>1.0</string>
+  <key>LSMinimumSystemVersion</key>
+  <string>${MACOSX_DEPLOYMENT_TARGET}</string>
+  <key>NSPrincipalClass</key>
+  <string>NSApplication</string>
+  <key>NSCameraUsageDescription</key>
+  <string>Camera access needed for video calling</string>
+  <key>NSMicrophoneUsageDescription</key>
+  <string>Microphone access needed for video calling</string>
+</dict>
+</plist>
\ No newline at end of file
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/main.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/main.m
new file mode 100644
index 0000000000..79b17f5492
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/mac/main.m
@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AppKit/AppKit.h>
+
+#import "APPRTCAppDelegate.h"
+
+int main(int argc, char* argv[]) {
+  @autoreleasepool {
+    [NSApplication sharedApplication];
+    APPRTCAppDelegate* delegate = [[APPRTCAppDelegate alloc] init];
+    [NSApp setDelegate:delegate];
+    [NSApp run];
+  }
+}
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDAppClient_xctest.mm b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDAppClient_xctest.mm
new file mode 100644
index 0000000000..2694e49914
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDAppClient_xctest.mm
@@ -0,0 +1,266 @@
+/*
+ *  Copyright 2014 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <OCMock/OCMock.h>
+#import <QuartzCore/CoreAnimation.h>
+#import <XCTest/XCTest.h>
+
+#include "rtc_base/ssl_adapter.h"
+
+#import "sdk/objc/api/peerconnection/RTCMediaConstraints.h"
+#import "sdk/objc/api/peerconnection/RTCPeerConnectionFactory.h"
+
+#import "ARDAppClient+Internal.h"
+#import "ARDJoinResponse+Internal.h"
+#import "ARDMessageResponse+Internal.h"
+#import "ARDSettingsModel.h"
+
+@interface ARDAppClientTest : XCTestCase
+@end
+
+@implementation ARDAppClientTest
+
+#pragma mark - Mock helpers
+
+- (id)mockRoomServerClientForRoomId:(NSString *)roomId
+                           clientId:(NSString *)clientId
+                        isInitiator:(BOOL)isInitiator
+                           messages:(NSArray *)messages
+                     messageHandler:
+    (void (^)(ARDSignalingMessage *))messageHandler {
+  id mockRoomServerClient =
+      [OCMockObject mockForProtocol:@protocol(ARDRoomServerClient)];
+
+  // Successful join response.
+  ARDJoinResponse *joinResponse = [[ARDJoinResponse alloc] init];
+  joinResponse.result = kARDJoinResultTypeSuccess;
+  joinResponse.roomId = roomId;
+  joinResponse.clientId = clientId;
+  joinResponse.isInitiator = isInitiator;
+  joinResponse.messages = messages;
+
+  // Successful message response.
+  ARDMessageResponse *messageResponse = [[ARDMessageResponse alloc] init];
+  messageResponse.result = kARDMessageResultTypeSuccess;
+
+  // Return join response from above on join.
+  [[[mockRoomServerClient stub] andDo:^(NSInvocation *invocation) {
+    __unsafe_unretained void (^completionHandler)(ARDJoinResponse *response,
+                                                  NSError *error);
+    [invocation getArgument:&completionHandler atIndex:4];
+    completionHandler(joinResponse, nil);
+  }] joinRoomWithRoomId:roomId isLoopback:NO completionHandler:[OCMArg any]];
+
+  // Return message response from above on join.
+  [[[mockRoomServerClient stub] andDo:^(NSInvocation *invocation) {
+    __unsafe_unretained ARDSignalingMessage *message;
+    __unsafe_unretained void (^completionHandler)(ARDMessageResponse *response,
+                                                  NSError *error);
+    [invocation getArgument:&message atIndex:2];
+    [invocation getArgument:&completionHandler atIndex:5];
+    messageHandler(message);
+    completionHandler(messageResponse, nil);
+  }] sendMessage:[OCMArg any]
+            forRoomId:roomId
+             clientId:clientId
+    completionHandler:[OCMArg any]];
+
+  // Do nothing on leave.
+  [[[mockRoomServerClient stub] andDo:^(NSInvocation *invocation) {
+    __unsafe_unretained void (^completionHandler)(NSError *error);
+    [invocation getArgument:&completionHandler atIndex:4];
+    if (completionHandler) {
+      completionHandler(nil);
+    }
+  }] leaveRoomWithRoomId:roomId
+                clientId:clientId
+       completionHandler:[OCMArg any]];
+
+  return mockRoomServerClient;
+}
+
+- (id)mockSignalingChannelForRoomId:(NSString *)roomId
+                           clientId:(NSString *)clientId
+                     messageHandler:
+    (void (^)(ARDSignalingMessage *message))messageHandler {
+  id mockSignalingChannel =
+      [OCMockObject niceMockForProtocol:@protocol(ARDSignalingChannel)];
+  [[mockSignalingChannel stub] registerForRoomId:roomId clientId:clientId];
+  [[[mockSignalingChannel stub] andDo:^(NSInvocation *invocation) {
+    __unsafe_unretained ARDSignalingMessage *message;
+    [invocation getArgument:&message atIndex:2];
+    messageHandler(message);
+  }] sendMessage:[OCMArg any]];
+  return mockSignalingChannel;
+}
+
+- (id)mockTURNClient {
+  id mockTURNClient =
+      [OCMockObject mockForProtocol:@protocol(ARDTURNClient)];
+  [[[mockTURNClient stub] andDo:^(NSInvocation *invocation) {
+    // Don't return anything in TURN response.
+    __unsafe_unretained void (^completionHandler)(NSArray *turnServers,
+                                                  NSError *error);
+    [invocation getArgument:&completionHandler atIndex:2];
+    completionHandler([NSArray array], nil);
+  }] requestServersWithCompletionHandler:[OCMArg any]];
+  return mockTURNClient;
+}
+
+- (id)mockSettingsModel {
+  ARDSettingsModel *model = [[ARDSettingsModel alloc] init];
+  id partialMock = [OCMockObject partialMockForObject:model];
+  [[[partialMock stub] andReturn:@[ @"640x480", @"960x540", @"1280x720" ]]
+      availableVideoResolutions];
+
+  return model;
+}
+
+- (ARDAppClient *)createAppClientForRoomId:(NSString *)roomId
+                                  clientId:(NSString *)clientId
+                               isInitiator:(BOOL)isInitiator
+                                  messages:(NSArray *)messages
+                            messageHandler:
+    (void (^)(ARDSignalingMessage *message))messageHandler
+                          connectedHandler:(void (^)(void))connectedHandler
+                    localVideoTrackHandler:(void (^)(void))localVideoTrackHandler {
+  id turnClient = [self mockTURNClient];
+  id signalingChannel = [self mockSignalingChannelForRoomId:roomId
+                                                   clientId:clientId
+                                             messageHandler:messageHandler];
+  id roomServerClient =
+      [self mockRoomServerClientForRoomId:roomId
+                                 clientId:clientId
+                              isInitiator:isInitiator
+                                 messages:messages
+                           messageHandler:messageHandler];
+  id delegate =
+      [OCMockObject niceMockForProtocol:@protocol(ARDAppClientDelegate)];
+  [[[delegate stub] andDo:^(NSInvocation *invocation) {
+    connectedHandler();
+  }] appClient:[OCMArg any]
+      didChangeConnectionState:RTCIceConnectionStateConnected];
+  [[[delegate stub] andDo:^(NSInvocation *invocation) {
+    localVideoTrackHandler();
+  }] appClient:[OCMArg any]
+      didReceiveLocalVideoTrack:[OCMArg any]];
+
+  return [[ARDAppClient alloc] initWithRoomServerClient:roomServerClient
+                                       signalingChannel:signalingChannel
+                                             turnClient:turnClient
+                                               delegate:delegate];
+}
+
+#pragma mark - Cases
+
+// Tests that an ICE connection is established between two ARDAppClient objects
+// where one is set up as a caller and the other the answerer. Network
+// components are mocked out and messages are relayed directly from object to
+// object. It's expected that both clients reach the
+// RTCIceConnectionStateConnected state within a reasonable amount of time.
+- (void)testSession {
+  // Need block arguments here because we're setting up a callbacks before we
+  // create the clients.
+  ARDAppClient *caller = nil;
+  ARDAppClient *answerer = nil;
+  __block __weak ARDAppClient *weakCaller = nil;
+  __block __weak ARDAppClient *weakAnswerer = nil;
+  NSString *roomId = @"testRoom";
+  NSString *callerId = @"testCallerId";
+  NSString *answererId = @"testAnswererId";
+
+  XCTestExpectation *callerConnectionExpectation =
+      [self expectationWithDescription:@"Caller PC connected"];
+  XCTestExpectation *answererConnectionExpectation =
+      [self expectationWithDescription:@"Answerer PC connected"];
+
+  caller = [self createAppClientForRoomId:roomId
+                                 clientId:callerId
+                              isInitiator:YES
+                                 messages:[NSArray array]
+                           messageHandler:^(ARDSignalingMessage *message) {
+    ARDAppClient *strongAnswerer = weakAnswerer;
+    [strongAnswerer channel:strongAnswerer.channel didReceiveMessage:message];
+  }                      connectedHandler:^{
+    [callerConnectionExpectation fulfill];
+  }                localVideoTrackHandler:^{
+  }];
+  // TODO(tkchin): Figure out why DTLS-SRTP constraint causes thread assertion
+  // crash in Debug.
+  caller.defaultPeerConnectionConstraints =
+      [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:nil
+                                                           optionalConstraints:nil];
+  weakCaller = caller;
+
+  answerer = [self createAppClientForRoomId:roomId
+                                   clientId:answererId
+                                isInitiator:NO
+                                   messages:[NSArray array]
+                             messageHandler:^(ARDSignalingMessage *message) {
+    ARDAppClient *strongCaller = weakCaller;
+    [strongCaller channel:strongCaller.channel didReceiveMessage:message];
+  }                        connectedHandler:^{
+    [answererConnectionExpectation fulfill];
+  }                  localVideoTrackHandler:^{
+  }];
+  // TODO(tkchin): Figure out why DTLS-SRTP constraint causes thread assertion
+  // crash in Debug.
+  answerer.defaultPeerConnectionConstraints =
+      [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:nil
+                                                           optionalConstraints:nil];
+  weakAnswerer = answerer;
+
+  // Kick off connection.
+  [caller connectToRoomWithId:roomId settings:[self mockSettingsModel] isLoopback:NO];
+  [answerer connectToRoomWithId:roomId settings:[self mockSettingsModel] isLoopback:NO];
+  [self waitForExpectationsWithTimeout:20 handler:^(NSError *error) {
+    if (error) {
+      XCTFail(@"Expectation failed with error %@.", error);
+    }
+  }];
+}
+
+// Test to see that we get a local video connection
+// Note this will currently pass even when no camera is connected as a local
+// video track is created regardless (Perhaps there should be a test for that...)
+#if !TARGET_IPHONE_SIMULATOR // Expect to fail on simulator due to no camera support
+- (void)testSessionShouldGetLocalVideoTrackCallback {
+  ARDAppClient *caller = nil;
+  NSString *roomId = @"testRoom";
+  NSString *callerId = @"testCallerId";
+
+  XCTestExpectation *localVideoTrackExpectation =
+      [self expectationWithDescription:@"Caller got local video."];
+
+  caller = [self createAppClientForRoomId:roomId
+                                 clientId:callerId
+                              isInitiator:YES
+                                 messages:[NSArray array]
+                           messageHandler:^(ARDSignalingMessage *message) {}
+                         connectedHandler:^{}
+                   localVideoTrackHandler:^{ [localVideoTrackExpectation fulfill]; }];
+  caller.defaultPeerConnectionConstraints =
+      [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:nil
+                                                           optionalConstraints:nil];
+
+  // Kick off connection.
+  [caller connectToRoomWithId:roomId
+                     settings:[self mockSettingsModel]
+                   isLoopback:NO];
+  [self waitForExpectationsWithTimeout:20 handler:^(NSError *error) {
+    if (error) {
+      XCTFail("Expectation timed out with error: %@.", error);
+    }
+  }];
+}
+#endif
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDFileCaptureController_xctest.mm b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDFileCaptureController_xctest.mm
new file mode 100644
index 0000000000..2e39834190
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDFileCaptureController_xctest.mm
@@ -0,0 +1,62 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <OCMock/OCMock.h>
+#import <XCTest/XCTest.h>
+
+#import "ARDFileCaptureController.h"
+
+#import "sdk/objc/components/capturer/RTCFileVideoCapturer.h"
+
+NS_CLASS_AVAILABLE_IOS(10)
+@interface ARDFileCaptureControllerTests : XCTestCase
+
+@property(nonatomic, strong) ARDFileCaptureController *fileCaptureController;
+@property(nonatomic, strong) id fileCapturerMock;
+
+@end
+
+@implementation ARDFileCaptureControllerTests
+
+@synthesize fileCaptureController = _fileCaptureController;
+@synthesize fileCapturerMock = _fileCapturerMock;
+
+- (void)setUp {
+  [super setUp];
+  self.fileCapturerMock = OCMClassMock([RTC_OBJC_TYPE(RTCFileVideoCapturer) class]);
+  self.fileCaptureController =
+      [[ARDFileCaptureController alloc] initWithCapturer:self.fileCapturerMock];
+}
+
+- (void)tearDown {
+  self.fileCaptureController = nil;
+  [self.fileCapturerMock stopMocking];
+  self.fileCapturerMock = nil;
+  [super tearDown];
+}
+
+- (void)testCaptureIsStarted {
+  [[self.fileCapturerMock expect] startCapturingFromFileNamed:[OCMArg any] onError:[OCMArg any]];
+
+  [self.fileCaptureController startCapture];
+
+  [self.fileCapturerMock verify];
+}
+
+- (void)testCaptureIsStoped {
+  [[self.fileCapturerMock expect] stopCapture];
+
+  [self.fileCaptureController stopCapture];
+
+  [self.fileCapturerMock verify];
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm
new file mode 100644
index 0000000000..dc62798963
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm
@@ -0,0 +1,96 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <OCMock/OCMock.h>
+#import <XCTest/XCTest.h>
+
+#import "sdk/objc/api/peerconnection/RTCMediaConstraints.h"
+
+#import "ARDSettingsModel+Private.h"
+#import "ARDSettingsStore.h"
+
+
+@interface ARDSettingsModelTests : XCTestCase {
+  ARDSettingsModel *_model;
+}
+@end
+
+@implementation ARDSettingsModelTests
+
+- (id)setupMockStore {
+  id storeMock = [OCMockObject mockForClass:[ARDSettingsStore class]];
+
+  id partialMock = [OCMockObject partialMockForObject:_model];
+  [[[partialMock stub] andReturn:storeMock] settingsStore];
+  [[[partialMock stub] andReturn:@[ @"640x480", @"960x540", @"1280x720" ]]
+      availableVideoResolutions];
+
+  return storeMock;
+}
+
+- (void)setUp {
+  _model = [[ARDSettingsModel alloc] init];
+}
+
+- (void)testRetrievingSetting {
+  id storeMock = [self setupMockStore];
+  [[[storeMock expect] andReturn:@"640x480"] videoResolution];
+  NSString *string = [_model currentVideoResolutionSettingFromStore];
+
+  XCTAssertEqualObjects(string, @"640x480");
+}
+
+- (void)testStoringInvalidConstraintReturnsNo {
+  id storeMock = [self setupMockStore];
+  [([[storeMock stub] andReturn:@"960x480"])videoResolution];
+  XCTAssertFalse([_model storeVideoResolutionSetting:@"960x480"]);
+}
+
+- (void)testWidthConstraintFromStore {
+  id storeMock = [self setupMockStore];
+  [([[storeMock stub] andReturn:@"1270x480"])videoResolution];
+  int width = [_model currentVideoResolutionWidthFromStore];
+
+  XCTAssertEqual(width, 1270);
+}
+
+- (void)testHeightConstraintFromStore {
+  id storeMock = [self setupMockStore];
+  [([[storeMock stub] andReturn:@"960x540"])videoResolution];
+  int height = [_model currentVideoResolutionHeightFromStore];
+
+  XCTAssertEqual(height, 540);
+}
+
+- (void)testConstraintComponentIsNilWhenInvalidConstraintString {
+  id storeMock = [self setupMockStore];
+  [([[storeMock stub] andReturn:@"invalid"])videoResolution];
+  int width = [_model currentVideoResolutionWidthFromStore];
+
+  XCTAssertEqual(width, 0);
+}
+
+- (void)testStoringAudioSetting {
+  id storeMock = [self setupMockStore];
+  [[storeMock expect] setAudioOnly:YES];
+
+  [_model storeAudioOnlySetting:YES];
+  [storeMock verify];
+}
+
+- (void)testReturningDefaultCallOption {
+  id storeMock = [self setupMockStore];
+  [[[storeMock stub] andReturnValue:@YES] useManualAudioConfig];
+
+  XCTAssertTrue([_model currentUseManualAudioConfigSettingFromStore]);
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/main.mm b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/main.mm
new file mode 100644
index 0000000000..3625ffd7bf
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/tests/main.mm
@@ -0,0 +1,21 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+#include "test/ios/coverage_util_ios.h"
+
+int main(int argc, char* argv[]) {
+  rtc::test::ConfigureCoverageReportPath();
+
+  @autoreleasepool {
+    return UIApplicationMain(argc, argv, nil, nil);
+  }
+}
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/LICENSE b/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/LICENSE
new file mode 100644
index 0000000000..c01a79c3bd
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/LICENSE
@@ -0,0 +1,15 @@
+
+   Copyright 2012 Square Inc.
+
+   Licensed under the Apache License, Version 2.0 (the "License");
+   you may not use this file except in compliance with the License.
+   You may obtain a copy of the License at
+
+       http://www.apache.org/licenses/LICENSE-2.0
+
+   Unless required by applicable law or agreed to in writing, software
+   distributed under the License is distributed on an "AS IS" BASIS,
+   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+   See the License for the specific language governing permissions and
+   limitations under the License.
+
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.h b/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.h
new file mode 100644
index 0000000000..a230646073
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.h
@@ -0,0 +1,135 @@
+//
+//   Copyright 2012 Square Inc.
+//
+//   Licensed under the Apache License, Version 2.0 (the "License");
+//   you may not use this file except in compliance with the License.
+//   You may obtain a copy of the License at
+//
+//       http://www.apache.org/licenses/LICENSE-2.0
+//
+//   Unless required by applicable law or agreed to in writing, software
+//   distributed under the License is distributed on an "AS IS" BASIS,
+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+//   See the License for the specific language governing permissions and
+//   limitations under the License.
+//
+
+#import <Foundation/Foundation.h>
+#import <Security/SecCertificate.h>
+
+typedef enum {
+  SR_CONNECTING = 0,
+  SR_OPEN = 1,
+  SR_CLOSING = 2,
+  SR_CLOSED = 3,
+} SRReadyState;
+
+typedef enum SRStatusCode : NSInteger {
+  SRStatusCodeNormal = 1000,
+  SRStatusCodeGoingAway = 1001,
+  SRStatusCodeProtocolError = 1002,
+  SRStatusCodeUnhandledType = 1003,
+  // 1004 reserved.
+  SRStatusNoStatusReceived = 1005,
+  // 1004-1006 reserved.
+  SRStatusCodeInvalidUTF8 = 1007,
+  SRStatusCodePolicyViolated = 1008,
+  SRStatusCodeMessageTooBig = 1009,
+} SRStatusCode;
+
+@class SRWebSocket;
+
+extern NSString *const SRWebSocketErrorDomain;
+extern NSString *const SRHTTPResponseErrorKey;
+
+#pragma mark - SRWebSocketDelegate
+
+@protocol SRWebSocketDelegate;
+
+#pragma mark - SRWebSocket
+
+@interface SRWebSocket : NSObject <NSStreamDelegate>
+
+@property(nonatomic, weak) id<SRWebSocketDelegate> delegate;
+
+@property(nonatomic, readonly) SRReadyState readyState;
+@property(nonatomic, readonly, retain) NSURL *url;
+
+// This returns the negotiated protocol.
+// It will be nil until after the handshake completes.
+@property(nonatomic, readonly, copy) NSString *protocol;
+
+// Protocols should be an array of strings that turn into Sec-WebSocket-Protocol.
+- (id)initWithURLRequest:(NSURLRequest *)request protocols:(NSArray *)protocols;
+- (id)initWithURLRequest:(NSURLRequest *)request;
+
+// Some helper constructors.
+- (id)initWithURL:(NSURL *)url protocols:(NSArray *)protocols;
+- (id)initWithURL:(NSURL *)url;
+
+// Delegate queue will be dispatch_main_queue by default.
+// You cannot set both OperationQueue and dispatch_queue.
+- (void)setDelegateOperationQueue:(NSOperationQueue *)queue;
+- (void)setDelegateDispatchQueue:(dispatch_queue_t)queue;
+
+// By default, it will schedule itself on +[NSRunLoop SR_networkRunLoop] using defaultModes.
+- (void)scheduleInRunLoop:(NSRunLoop *)aRunLoop forMode:(NSString *)mode;
+- (void)unscheduleFromRunLoop:(NSRunLoop *)aRunLoop forMode:(NSString *)mode;
+
+// SRWebSockets are intended for one-time-use only.  Open should be called once and only once.
+- (void)open;
+
+- (void)close;
+- (void)closeWithCode:(NSInteger)code reason:(NSString *)reason;
+
+// Send a UTF8 String or Data.
+- (void)send:(id)data;
+
+// Send Data (can be nil) in a ping message.
+- (void)sendPing:(NSData *)data;
+
+@end
+
+#pragma mark - SRWebSocketDelegate
+
+@protocol SRWebSocketDelegate <NSObject>
+
+// message will either be an NSString if the server is using text
+// or NSData if the server is using binary.
+- (void)webSocket:(SRWebSocket *)webSocket didReceiveMessage:(id)message;
+
+@optional
+
+- (void)webSocketDidOpen:(SRWebSocket *)webSocket;
+- (void)webSocket:(SRWebSocket *)webSocket didFailWithError:(NSError *)error;
+- (void)webSocket:(SRWebSocket *)webSocket
+    didCloseWithCode:(NSInteger)code
+              reason:(NSString *)reason
+            wasClean:(BOOL)wasClean;
+- (void)webSocket:(SRWebSocket *)webSocket didReceivePong:(NSData *)pongPayload;
+
+@end
+
+#pragma mark - NSURLRequest (CertificateAdditions)
+
+@interface NSURLRequest (CertificateAdditions)
+
+@property(nonatomic, retain, readonly) NSArray *SR_SSLPinnedCertificates;
+
+@end
+
+#pragma mark - NSMutableURLRequest (CertificateAdditions)
+
+@interface NSMutableURLRequest (CertificateAdditions)
+
+@property(nonatomic, retain) NSArray *SR_SSLPinnedCertificates;
+
+@end
+
+#pragma mark - NSRunLoop (SRWebSocket)
+
+@interface NSRunLoop (SRWebSocket)
+
++ (NSRunLoop *)SR_networkRunLoop;
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m b/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m
new file mode 100644
index 0000000000..ab0d1b89bc
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m
@@ -0,0 +1,1774 @@
+//
+//   Copyright 2012 Square Inc.
+//
+//   Licensed under the Apache License, Version 2.0 (the "License");
+//   you may not use this file except in compliance with the License.
+//   You may obtain a copy of the License at
+//
+//       http://www.apache.org/licenses/LICENSE-2.0
+//
+//   Unless required by applicable law or agreed to in writing, software
+//   distributed under the License is distributed on an "AS IS" BASIS,
+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+//   See the License for the specific language governing permissions and
+//   limitations under the License.
+//
+
+
+#import "SRWebSocket.h"
+
+#if TARGET_OS_IPHONE
+#define HAS_ICU
+#endif
+
+#ifdef HAS_ICU
+#import <unicode/utf8.h>
+#endif
+
+#if TARGET_OS_IPHONE
+#import <Endian.h>
+#else
+#import <CoreServices/CoreServices.h>
+#endif
+
+#import <CommonCrypto/CommonDigest.h>
+#import <Security/SecRandom.h>
+
+#if OS_OBJECT_USE_OBJC_RETAIN_RELEASE
+#define sr_dispatch_retain(x)
+#define sr_dispatch_release(x)
+#define maybe_bridge(x) ((__bridge void *) x)
+#else
+#define sr_dispatch_retain(x) dispatch_retain(x)
+#define sr_dispatch_release(x) dispatch_release(x)
+#define maybe_bridge(x) (x)
+#endif
+
+#if !__has_feature(objc_arc) 
+#error SocketRocket must be compiled with ARC enabled
+#endif
+
+
+typedef enum  {
+    SROpCodeTextFrame = 0x1,
+    SROpCodeBinaryFrame = 0x2,
+    // 3-7 reserved.
+    SROpCodeConnectionClose = 0x8,
+    SROpCodePing = 0x9,
+    SROpCodePong = 0xA,
+    // B-F reserved.
+} SROpCode;
+
+typedef struct {
+    BOOL fin;
+//  BOOL rsv1;
+//  BOOL rsv2;
+//  BOOL rsv3;
+    uint8_t opcode;
+    BOOL masked;
+    uint64_t payload_length;
+} frame_header;
+
+static NSString *const SRWebSocketAppendToSecKeyString = @"258EAFA5-E914-47DA-95CA-C5AB0DC85B11";
+
+static inline int32_t validate_dispatch_data_partial_string(NSData *data);
+static inline void SRFastLog(NSString *format, ...);
+
+@interface NSData (SRWebSocket)
+
+- (NSString *)stringBySHA1ThenBase64Encoding;
+
+@end
+
+
+@interface NSString (SRWebSocket)
+
+- (NSString *)stringBySHA1ThenBase64Encoding;
+
+@end
+
+
+@interface NSURL (SRWebSocket)
+
+// The origin isn't really applicable for a native application.
+// So instead, just map ws -> http and wss -> https.
+- (NSString *)SR_origin;
+
+@end
+
+
+@interface _SRRunLoopThread : NSThread
+
+@property (nonatomic, readonly) NSRunLoop *runLoop;
+
+@end
+
+
+static NSString *newSHA1String(const char *bytes, size_t length) {
+    uint8_t md[CC_SHA1_DIGEST_LENGTH];
+
+    assert(length >= 0);
+    assert(length <= UINT32_MAX);
+    CC_SHA1(bytes, (CC_LONG)length, md);
+    
+    NSData *data = [NSData dataWithBytes:md length:CC_SHA1_DIGEST_LENGTH];
+    
+    if ([data respondsToSelector:@selector(base64EncodedStringWithOptions:)]) {
+        return [data base64EncodedStringWithOptions:0];
+    }
+    
+    return [data base64Encoding];
+}
+
+@implementation NSData (SRWebSocket)
+
+- (NSString *)stringBySHA1ThenBase64Encoding;
+{
+    return newSHA1String(self.bytes, self.length);
+}
+
+@end
+
+
+@implementation NSString (SRWebSocket)
+
+- (NSString *)stringBySHA1ThenBase64Encoding;
+{
+    return newSHA1String(self.UTF8String, self.length);
+}
+
+@end
+
+NSString *const SRWebSocketErrorDomain = @"SRWebSocketErrorDomain";
+NSString *const SRHTTPResponseErrorKey = @"HTTPResponseStatusCode";
+
+// Returns number of bytes consumed. Returning 0 means you didn't match.
+// Sends bytes to callback handler;
+typedef size_t (^stream_scanner)(NSData *collected_data);
+
+typedef void (^data_callback)(SRWebSocket *webSocket,  NSData *data);
+
+@interface SRIOConsumer : NSObject {
+    stream_scanner _scanner;
+    data_callback _handler;
+    size_t _bytesNeeded;
+    BOOL _readToCurrentFrame;
+    BOOL _unmaskBytes;
+}
+@property (nonatomic, copy, readonly) stream_scanner consumer;
+@property (nonatomic, copy, readonly) data_callback handler;
+@property (nonatomic, assign) size_t bytesNeeded;
+@property (nonatomic, assign, readonly) BOOL readToCurrentFrame;
+@property (nonatomic, assign, readonly) BOOL unmaskBytes;
+
+@end
+
+// This class is not thread-safe, and is expected to always be run on the same queue.
+@interface SRIOConsumerPool : NSObject
+
+- (id)initWithBufferCapacity:(NSUInteger)poolSize;
+
+- (SRIOConsumer *)consumerWithScanner:(stream_scanner)scanner handler:(data_callback)handler bytesNeeded:(size_t)bytesNeeded readToCurrentFrame:(BOOL)readToCurrentFrame unmaskBytes:(BOOL)unmaskBytes;
+- (void)returnConsumer:(SRIOConsumer *)consumer;
+
+@end
+
+@interface SRWebSocket ()  <NSStreamDelegate>
+
+- (void)_writeData:(NSData *)data;
+- (void)_closeWithProtocolError:(NSString *)message;
+- (void)_failWithError:(NSError *)error;
+
+- (void)_disconnect;
+
+- (void)_readFrameNew;
+- (void)_readFrameContinue;
+
+- (void)_pumpScanner;
+
+- (void)_pumpWriting;
+
+- (void)_addConsumerWithScanner:(stream_scanner)consumer callback:(data_callback)callback;
+- (void)_addConsumerWithDataLength:(size_t)dataLength callback:(data_callback)callback readToCurrentFrame:(BOOL)readToCurrentFrame unmaskBytes:(BOOL)unmaskBytes;
+- (void)_addConsumerWithScanner:(stream_scanner)consumer callback:(data_callback)callback dataLength:(size_t)dataLength;
+- (void)_readUntilBytes:(const void *)bytes length:(size_t)length callback:(data_callback)dataHandler;
+- (void)_readUntilHeaderCompleteWithCallback:(data_callback)dataHandler;
+
+- (void)_sendFrameWithOpcode:(SROpCode)opcode data:(id)data;
+
+- (BOOL)_checkHandshake:(CFHTTPMessageRef)httpMessage;
+- (void)_SR_commonInit;
+
+- (void)_initializeStreams;
+- (void)_connect;
+
+@property (nonatomic) SRReadyState readyState;
+
+@property (nonatomic) NSOperationQueue *delegateOperationQueue;
+@property (nonatomic) dispatch_queue_t delegateDispatchQueue;
+
+@end
+
+
+@implementation SRWebSocket {
+    NSInteger _webSocketVersion;
+    
+    NSOperationQueue *_delegateOperationQueue;
+    dispatch_queue_t _delegateDispatchQueue;
+    
+    dispatch_queue_t _workQueue;
+    NSMutableArray *_consumers;
+
+    NSInputStream *_inputStream;
+    NSOutputStream *_outputStream;
+   
+    NSMutableData *_readBuffer;
+    NSUInteger _readBufferOffset;
+ 
+    NSMutableData *_outputBuffer;
+    NSUInteger _outputBufferOffset;
+
+    uint8_t _currentFrameOpcode;
+    size_t _currentFrameCount;
+    size_t _readOpCount;
+    uint32_t _currentStringScanPosition;
+    NSMutableData *_currentFrameData;
+    
+    NSString *_closeReason;
+    
+    NSString *_secKey;
+    
+    BOOL _pinnedCertFound;
+    
+    uint8_t _currentReadMaskKey[4];
+    size_t _currentReadMaskOffset;
+
+    BOOL _consumerStopped;
+    
+    BOOL _closeWhenFinishedWriting;
+    BOOL _failed;
+
+    BOOL _secure;
+    NSURLRequest *_urlRequest;
+
+    CFHTTPMessageRef _receivedHTTPHeaders;
+    
+    BOOL _sentClose;
+    BOOL _didFail;
+    int _closeCode;
+    
+    BOOL _isPumping;
+    
+    NSMutableSet *_scheduledRunloops;
+    
+    // We use this to retain ourselves.
+    __strong SRWebSocket *_selfRetain;
+    
+    NSArray *_requestedProtocols;
+    SRIOConsumerPool *_consumerPool;
+}
+
+@synthesize delegate = _delegate;
+@synthesize url = _url;
+@synthesize readyState = _readyState;
+@synthesize protocol = _protocol;
+
+static __strong NSData *CRLFCRLF;
+
++ (void)initialize;
+{
+    CRLFCRLF = [[NSData alloc] initWithBytes:"\r\n\r\n" length:4];
+}
+
+- (id)initWithURLRequest:(NSURLRequest *)request protocols:(NSArray *)protocols;
+{
+    self = [super init];
+    if (self) {
+        assert(request.URL);
+        _url = request.URL;
+        _urlRequest = request;
+        
+        _requestedProtocols = [protocols copy];
+        
+        [self _SR_commonInit];
+    }
+    
+    return self;
+}
+
+- (id)initWithURLRequest:(NSURLRequest *)request;
+{
+    return [self initWithURLRequest:request protocols:nil];
+}
+
+- (id)initWithURL:(NSURL *)url;
+{
+    return [self initWithURL:url protocols:nil];
+}
+
+- (id)initWithURL:(NSURL *)url protocols:(NSArray *)protocols;
+{
+    NSMutableURLRequest *request = [[NSMutableURLRequest alloc] initWithURL:url];    
+    return [self initWithURLRequest:request protocols:protocols];
+}
+
+- (void)_SR_commonInit;
+{
+    
+    NSString *scheme = _url.scheme.lowercaseString;
+    assert([scheme isEqualToString:@"ws"] || [scheme isEqualToString:@"http"] || [scheme isEqualToString:@"wss"] || [scheme isEqualToString:@"https"]);
+    
+    if ([scheme isEqualToString:@"wss"] || [scheme isEqualToString:@"https"]) {
+        _secure = YES;
+    }
+    
+    _readyState = SR_CONNECTING;
+    _consumerStopped = YES;
+    _webSocketVersion = 13;
+    
+    _workQueue = dispatch_queue_create(NULL, DISPATCH_QUEUE_SERIAL);
+    
+    // Going to set a specific on the queue so we can validate we're on the work queue
+    dispatch_queue_set_specific(_workQueue, (__bridge void *)self, maybe_bridge(_workQueue), NULL);
+    
+    _delegateDispatchQueue = dispatch_get_main_queue();
+    sr_dispatch_retain(_delegateDispatchQueue);
+    
+    _readBuffer = [[NSMutableData alloc] init];
+    _outputBuffer = [[NSMutableData alloc] init];
+    
+    _currentFrameData = [[NSMutableData alloc] init];
+
+    _consumers = [[NSMutableArray alloc] init];
+    
+    _consumerPool = [[SRIOConsumerPool alloc] init];
+    
+    _scheduledRunloops = [[NSMutableSet alloc] init];
+    
+    [self _initializeStreams];
+    
+    // default handlers
+}
+
+- (void)assertOnWorkQueue;
+{
+    assert(dispatch_get_specific((__bridge void *)self) == maybe_bridge(_workQueue));
+}
+
+- (void)dealloc
+{
+    _inputStream.delegate = nil;
+    _outputStream.delegate = nil;
+
+    [_inputStream close];
+    [_outputStream close];
+    
+    sr_dispatch_release(_workQueue);
+    _workQueue = NULL;
+    
+    if (_receivedHTTPHeaders) {
+        CFRelease(_receivedHTTPHeaders);
+        _receivedHTTPHeaders = NULL;
+    }
+    
+    if (_delegateDispatchQueue) {
+        sr_dispatch_release(_delegateDispatchQueue);
+        _delegateDispatchQueue = NULL;
+    }
+}
+
+#ifndef NDEBUG
+
+- (void)setReadyState:(SRReadyState)aReadyState;
+{
+    [self willChangeValueForKey:@"readyState"];
+    assert(aReadyState > _readyState);
+    _readyState = aReadyState;
+    [self didChangeValueForKey:@"readyState"];
+}
+
+#endif
+
+- (void)open;
+{
+    assert(_url);
+    NSAssert(_readyState == SR_CONNECTING, @"Cannot call -(void)open on SRWebSocket more than once");
+
+    _selfRetain = self;
+    
+    [self _connect];
+}
+
+// Calls block on delegate queue
+- (void)_performDelegateBlock:(dispatch_block_t)block;
+{
+    if (_delegateOperationQueue) {
+        [_delegateOperationQueue addOperationWithBlock:block];
+    } else {
+        assert(_delegateDispatchQueue);
+        dispatch_async(_delegateDispatchQueue, block);
+    }
+}
+
+- (void)setDelegateDispatchQueue:(dispatch_queue_t)queue;
+{
+    if (queue) {
+        sr_dispatch_retain(queue);
+    }
+    
+    if (_delegateDispatchQueue) {
+        sr_dispatch_release(_delegateDispatchQueue);
+    }
+    
+    _delegateDispatchQueue = queue;
+}
+
+- (BOOL)_checkHandshake:(CFHTTPMessageRef)httpMessage;
+{
+    NSString *acceptHeader = CFBridgingRelease(CFHTTPMessageCopyHeaderFieldValue(httpMessage, CFSTR("Sec-WebSocket-Accept")));
+
+    if (acceptHeader == nil) {
+        return NO;
+    }
+    
+    NSString *concattedString = [_secKey stringByAppendingString:SRWebSocketAppendToSecKeyString];
+    NSString *expectedAccept = [concattedString stringBySHA1ThenBase64Encoding];
+    
+    return [acceptHeader isEqualToString:expectedAccept];
+}
+
+- (void)_HTTPHeadersDidFinish;
+{
+    NSInteger responseCode = CFHTTPMessageGetResponseStatusCode(_receivedHTTPHeaders);
+    
+    if (responseCode >= 400) {
+        SRFastLog(@"Request failed with response code %d", responseCode);
+        [self _failWithError:[NSError errorWithDomain:SRWebSocketErrorDomain code:2132 userInfo:@{NSLocalizedDescriptionKey:[NSString stringWithFormat:@"received bad response code from server %ld", (long)responseCode], SRHTTPResponseErrorKey:@(responseCode)}]];
+        return;
+    }
+    
+    if(![self _checkHandshake:_receivedHTTPHeaders]) {
+        [self _failWithError:[NSError errorWithDomain:SRWebSocketErrorDomain code:2133 userInfo:[NSDictionary dictionaryWithObject:[NSString stringWithFormat:@"Invalid Sec-WebSocket-Accept response"] forKey:NSLocalizedDescriptionKey]]];
+        return;
+    }
+    
+    NSString *negotiatedProtocol = CFBridgingRelease(CFHTTPMessageCopyHeaderFieldValue(_receivedHTTPHeaders, CFSTR("Sec-WebSocket-Protocol")));
+    if (negotiatedProtocol) {
+        // Make sure we requested the protocol
+        if ([_requestedProtocols indexOfObject:negotiatedProtocol] == NSNotFound) {
+            [self _failWithError:[NSError errorWithDomain:SRWebSocketErrorDomain code:2133 userInfo:[NSDictionary dictionaryWithObject:[NSString stringWithFormat:@"Server specified Sec-WebSocket-Protocol that wasn't requested"] forKey:NSLocalizedDescriptionKey]]];
+            return;
+        }
+        
+        _protocol = negotiatedProtocol;
+    }
+    
+    self.readyState = SR_OPEN;
+    
+    if (!_didFail) {
+        [self _readFrameNew];
+    }
+
+    [self _performDelegateBlock:^{
+        if ([self.delegate respondsToSelector:@selector(webSocketDidOpen:)]) {
+            [self.delegate webSocketDidOpen:self];
+        };
+    }];
+}
+
+
+- (void)_readHTTPHeader;
+{
+    if (_receivedHTTPHeaders == NULL) {
+        _receivedHTTPHeaders = CFHTTPMessageCreateEmpty(NULL, NO);
+    }
+
+    [self _readUntilHeaderCompleteWithCallback:^(SRWebSocket *self, NSData *data) {
+      CFHTTPMessageAppendBytes(self->_receivedHTTPHeaders, (const UInt8 *)data.bytes, data.length);
+
+      if (CFHTTPMessageIsHeaderComplete(self->_receivedHTTPHeaders)) {
+        SRFastLog(@"Finished reading headers %@",
+                  CFBridgingRelease(CFHTTPMessageCopyAllHeaderFields(self->_receivedHTTPHeaders)));
+        [self _HTTPHeadersDidFinish];
+      } else {
+        [self _readHTTPHeader];
+      }
+    }];
+}
+
+- (void)didConnect
+{
+    SRFastLog(@"Connected");
+    CFHTTPMessageRef request = CFHTTPMessageCreateRequest(NULL, CFSTR("GET"), (__bridge CFURLRef)_url, kCFHTTPVersion1_1);
+    
+    // Set host first so it defaults
+    CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Host"), (__bridge CFStringRef)(_url.port ? [NSString stringWithFormat:@"%@:%@", _url.host, _url.port] : _url.host));
+        
+    NSMutableData *keyBytes = [[NSMutableData alloc] initWithLength:16];
+    BOOL success = !SecRandomCopyBytes(kSecRandomDefault, keyBytes.length, keyBytes.mutableBytes);
+    assert(success);
+    
+    if ([keyBytes respondsToSelector:@selector(base64EncodedStringWithOptions:)]) {
+        _secKey = [keyBytes base64EncodedStringWithOptions:0];
+    } else {
+        _secKey = [keyBytes base64Encoding];
+    }
+    
+    assert([_secKey length] == 24);
+    
+    CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Upgrade"), CFSTR("websocket"));
+    CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Connection"), CFSTR("Upgrade"));
+    CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Sec-WebSocket-Key"), (__bridge CFStringRef)_secKey);
+    CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Sec-WebSocket-Version"), (__bridge CFStringRef)[NSString stringWithFormat:@"%ld", (long)_webSocketVersion]);
+    
+    CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Origin"), (__bridge CFStringRef)_url.SR_origin);
+    
+    if (_requestedProtocols) {
+        CFHTTPMessageSetHeaderFieldValue(request, CFSTR("Sec-WebSocket-Protocol"), (__bridge CFStringRef)[_requestedProtocols componentsJoinedByString:@", "]);
+    }
+
+    [_urlRequest.allHTTPHeaderFields enumerateKeysAndObjectsUsingBlock:^(id key, id obj, BOOL *stop) {
+        CFHTTPMessageSetHeaderFieldValue(request, (__bridge CFStringRef)key, (__bridge CFStringRef)obj);
+    }];
+
+    NSData *message = CFBridgingRelease(CFHTTPMessageCopySerializedMessage(request));
+    
+    CFRelease(request);
+
+    [self _writeData:message];
+    [self _readHTTPHeader];
+}
+
+- (void)_initializeStreams;
+{
+    assert(_url.port.unsignedIntValue <= UINT32_MAX);
+    uint32_t port = _url.port.unsignedIntValue;
+    if (port == 0) {
+        if (!_secure) {
+            port = 80;
+        } else {
+            port = 443;
+        }
+    }
+    NSString *host = _url.host;
+    
+    CFReadStreamRef readStream = NULL;
+    CFWriteStreamRef writeStream = NULL;
+    
+    CFStreamCreatePairWithSocketToHost(NULL, (__bridge CFStringRef)host, port, &readStream, &writeStream);
+    
+    _outputStream = CFBridgingRelease(writeStream);
+    _inputStream = CFBridgingRelease(readStream);
+    
+    
+    if (_secure) {
+        NSMutableDictionary *SSLOptions = [[NSMutableDictionary alloc] init];
+        
+        [_outputStream setProperty:(__bridge id)kCFStreamSocketSecurityLevelNegotiatedSSL forKey:(__bridge id)kCFStreamPropertySocketSecurityLevel];
+        
+        // If we're using pinned certs, don't validate the certificate chain
+        if ([_urlRequest SR_SSLPinnedCertificates].count) {
+            [SSLOptions setValue:[NSNumber numberWithBool:NO] forKey:(__bridge id)kCFStreamSSLValidatesCertificateChain];
+        }
+        
+#ifdef DEBUG
+        [SSLOptions setValue:[NSNumber numberWithBool:NO] forKey:(__bridge id)kCFStreamSSLValidatesCertificateChain];
+        NSLog(@"SocketRocket: In debug mode.  Allowing connection to any root cert");
+#endif
+        
+        [_outputStream setProperty:SSLOptions
+                            forKey:(__bridge id)kCFStreamPropertySSLSettings];
+    }
+    
+    _inputStream.delegate = self;
+    _outputStream.delegate = self;
+}
+
+- (void)_connect;
+{
+    if (!_scheduledRunloops.count) {
+        [self scheduleInRunLoop:[NSRunLoop SR_networkRunLoop] forMode:NSDefaultRunLoopMode];
+    }
+    
+    
+    [_outputStream open];
+    [_inputStream open];
+}
+
+- (void)scheduleInRunLoop:(NSRunLoop *)aRunLoop forMode:(NSString *)mode;
+{
+    [_outputStream scheduleInRunLoop:aRunLoop forMode:mode];
+    [_inputStream scheduleInRunLoop:aRunLoop forMode:mode];
+    
+    [_scheduledRunloops addObject:@[aRunLoop, mode]];
+}
+
+- (void)unscheduleFromRunLoop:(NSRunLoop *)aRunLoop forMode:(NSString *)mode;
+{
+    [_outputStream removeFromRunLoop:aRunLoop forMode:mode];
+    [_inputStream removeFromRunLoop:aRunLoop forMode:mode];
+    
+    [_scheduledRunloops removeObject:@[aRunLoop, mode]];
+}
+
+- (void)close;
+{
+    [self closeWithCode:SRStatusCodeNormal reason:nil];
+}
+
+- (void)closeWithCode:(NSInteger)code reason:(NSString *)reason;
+{
+    assert(code);
+    dispatch_async(_workQueue, ^{
+        if (self.readyState == SR_CLOSING || self.readyState == SR_CLOSED) {
+            return;
+        }
+        
+        BOOL wasConnecting = self.readyState == SR_CONNECTING;
+        
+        self.readyState = SR_CLOSING;
+        
+        SRFastLog(@"Closing with code %d reason %@", code, reason);
+        
+        if (wasConnecting) {
+            [self _disconnect];
+            return;
+        }
+
+        size_t maxMsgSize = [reason maximumLengthOfBytesUsingEncoding:NSUTF8StringEncoding];
+        NSMutableData *mutablePayload = [[NSMutableData alloc] initWithLength:sizeof(uint16_t) + maxMsgSize];
+        NSData *payload = mutablePayload;
+        
+        ((uint16_t *)mutablePayload.mutableBytes)[0] = EndianU16_BtoN(code);
+        
+        if (reason) {
+            NSRange remainingRange = {0};
+            
+            NSUInteger usedLength = 0;
+            
+            BOOL success = [reason getBytes:(char *)mutablePayload.mutableBytes + sizeof(uint16_t) maxLength:payload.length - sizeof(uint16_t) usedLength:&usedLength encoding:NSUTF8StringEncoding options:NSStringEncodingConversionExternalRepresentation range:NSMakeRange(0, reason.length) remainingRange:&remainingRange];
+            
+            assert(success);
+            assert(remainingRange.length == 0);
+
+            if (usedLength != maxMsgSize) {
+                payload = [payload subdataWithRange:NSMakeRange(0, usedLength + sizeof(uint16_t))];
+            }
+        }
+        
+        
+        [self _sendFrameWithOpcode:SROpCodeConnectionClose data:payload];
+    });
+}
+
+- (void)_closeWithProtocolError:(NSString *)message;
+{
+    // Need to shunt this on the _callbackQueue first to see if they received any messages 
+    [self _performDelegateBlock:^{
+        [self closeWithCode:SRStatusCodeProtocolError reason:message];
+        dispatch_async(self->_workQueue, ^{
+          [self _disconnect];
+        });
+    }];
+}
+
+- (void)_failWithError:(NSError *)error;
+{
+    dispatch_async(_workQueue, ^{
+        if (self.readyState != SR_CLOSED) {
+          self->_failed = YES;
+          [self _performDelegateBlock:^{
+            if ([self.delegate respondsToSelector:@selector(webSocket:didFailWithError:)]) {
+              [self.delegate webSocket:self didFailWithError:error];
+            }
+          }];
+
+          self.readyState = SR_CLOSED;
+          self->_selfRetain = nil;
+
+          SRFastLog(@"Failing with error %@", error.localizedDescription);
+
+          [self _disconnect];
+        }
+    });
+}
+
+- (void)_writeData:(NSData *)data;
+{    
+    [self assertOnWorkQueue];
+
+    if (_closeWhenFinishedWriting) {
+            return;
+    }
+    [_outputBuffer appendData:data];
+    [self _pumpWriting];
+}
+
+- (void)send:(id)data;
+{
+    NSAssert(self.readyState != SR_CONNECTING, @"Invalid State: Cannot call send: until connection is open");
+    // TODO: maybe not copy this for performance
+    data = [data copy];
+    dispatch_async(_workQueue, ^{
+        if ([data isKindOfClass:[NSString class]]) {
+            [self _sendFrameWithOpcode:SROpCodeTextFrame data:[(NSString *)data dataUsingEncoding:NSUTF8StringEncoding]];
+        } else if ([data isKindOfClass:[NSData class]]) {
+            [self _sendFrameWithOpcode:SROpCodeBinaryFrame data:data];
+        } else if (data == nil) {
+            [self _sendFrameWithOpcode:SROpCodeTextFrame data:data];
+        } else {
+            assert(NO);
+        }
+    });
+}
+
+- (void)sendPing:(NSData *)data;
+{
+    NSAssert(self.readyState == SR_OPEN, @"Invalid State: Cannot call send: until connection is open");
+    // TODO: maybe not copy this for performance
+    data = [data copy] ?: [NSData data]; // It's okay for a ping to be empty
+    dispatch_async(_workQueue, ^{
+        [self _sendFrameWithOpcode:SROpCodePing data:data];
+    });
+}
+
+- (void)handlePing:(NSData *)pingData;
+{
+    // Need to pingpong this off _callbackQueue first to make sure messages happen in order
+    [self _performDelegateBlock:^{
+      dispatch_async(self->_workQueue, ^{
+        [self _sendFrameWithOpcode:SROpCodePong data:pingData];
+      });
+    }];
+}
+
+- (void)handlePong:(NSData *)pongData;
+{
+    SRFastLog(@"Received pong");
+    [self _performDelegateBlock:^{
+        if ([self.delegate respondsToSelector:@selector(webSocket:didReceivePong:)]) {
+            [self.delegate webSocket:self didReceivePong:pongData];
+        }
+    }];
+}
+
+- (void)_handleMessage:(id)message
+{
+    SRFastLog(@"Received message");
+    [self _performDelegateBlock:^{
+        [self.delegate webSocket:self didReceiveMessage:message];
+    }];
+}
+
+
+static inline BOOL closeCodeIsValid(int closeCode) {
+    if (closeCode < 1000) {
+        return NO;
+    }
+    
+    if (closeCode >= 1000 && closeCode <= 1011) {
+        if (closeCode == 1004 ||
+            closeCode == 1005 ||
+            closeCode == 1006) {
+            return NO;
+        }
+        return YES;
+    }
+    
+    if (closeCode >= 3000 && closeCode <= 3999) {
+        return YES;
+    }
+    
+    if (closeCode >= 4000 && closeCode <= 4999) {
+        return YES;
+    }
+
+    return NO;
+}
+
+//  Note from RFC:
+//
+//  If there is a body, the first two
+//  bytes of the body MUST be a 2-byte unsigned integer (in network byte
+//  order) representing a status code with value /code/ defined in
+//  Section 7.4.  Following the 2-byte integer the body MAY contain UTF-8
+//  encoded data with value /reason/, the interpretation of which is not
+//  defined by this specification.
+
+- (void)handleCloseWithData:(NSData *)data;
+{
+    size_t dataSize = data.length;
+    __block uint16_t closeCode = 0;
+    
+    SRFastLog(@"Received close frame");
+    
+    if (dataSize == 1) {
+        // TODO handle error
+        [self _closeWithProtocolError:@"Payload for close must be larger than 2 bytes"];
+        return;
+    } else if (dataSize >= 2) {
+        [data getBytes:&closeCode length:sizeof(closeCode)];
+        _closeCode = EndianU16_BtoN(closeCode);
+        if (!closeCodeIsValid(_closeCode)) {
+            [self _closeWithProtocolError:[NSString stringWithFormat:@"Cannot have close code of %d", _closeCode]];
+            return;
+        }
+        if (dataSize > 2) {
+            _closeReason = [[NSString alloc] initWithData:[data subdataWithRange:NSMakeRange(2, dataSize - 2)] encoding:NSUTF8StringEncoding];
+            if (!_closeReason) {
+                [self _closeWithProtocolError:@"Close reason MUST be valid UTF-8"];
+                return;
+            }
+        }
+    } else {
+        _closeCode = SRStatusNoStatusReceived;
+    }
+    
+    [self assertOnWorkQueue];
+    
+    if (self.readyState == SR_OPEN) {
+        [self closeWithCode:1000 reason:nil];
+    }
+    dispatch_async(_workQueue, ^{
+        [self _disconnect];
+    });
+}
+
+- (void)_disconnect;
+{
+    [self assertOnWorkQueue];
+    SRFastLog(@"Trying to disconnect");
+    _closeWhenFinishedWriting = YES;
+    [self _pumpWriting];
+}
+
+- (void)_handleFrameWithData:(NSData *)frameData opCode:(NSInteger)opcode;
+{                
+    // Check that the current data is valid UTF8
+    
+    BOOL isControlFrame = (opcode == SROpCodePing || opcode == SROpCodePong || opcode == SROpCodeConnectionClose);
+    if (!isControlFrame) {
+        [self _readFrameNew];
+    } else {
+        dispatch_async(_workQueue, ^{
+            [self _readFrameContinue];
+        });
+    }
+    
+    switch (opcode) {
+        case SROpCodeTextFrame: {
+            NSString *str = [[NSString alloc] initWithData:frameData encoding:NSUTF8StringEncoding];
+            if (str == nil && frameData) {
+                [self closeWithCode:SRStatusCodeInvalidUTF8 reason:@"Text frames must be valid UTF-8"];
+                dispatch_async(_workQueue, ^{
+                    [self _disconnect];
+                });
+
+                return;
+            }
+            [self _handleMessage:str];
+            break;
+        }
+        case SROpCodeBinaryFrame:
+            [self _handleMessage:[frameData copy]];
+            break;
+        case SROpCodeConnectionClose:
+            [self handleCloseWithData:frameData];
+            break;
+        case SROpCodePing:
+            [self handlePing:frameData];
+            break;
+        case SROpCodePong:
+            [self handlePong:frameData];
+            break;
+        default:
+            [self _closeWithProtocolError:[NSString stringWithFormat:@"Unknown opcode %ld", (long)opcode]];
+            // TODO: Handle invalid opcode
+            break;
+    }
+}
+
+- (void)_handleFrameHeader:(frame_header)frame_header curData:(NSData *)curData;
+{
+    assert(frame_header.opcode != 0);
+    
+    if (self.readyState != SR_OPEN) {
+        return;
+    }
+    
+    
+    BOOL isControlFrame = (frame_header.opcode == SROpCodePing || frame_header.opcode == SROpCodePong || frame_header.opcode == SROpCodeConnectionClose);
+    
+    if (isControlFrame && !frame_header.fin) {
+        [self _closeWithProtocolError:@"Fragmented control frames not allowed"];
+        return;
+    }
+    
+    if (isControlFrame && frame_header.payload_length >= 126) {
+        [self _closeWithProtocolError:@"Control frames cannot have payloads larger than 126 bytes"];
+        return;
+    }
+    
+    if (!isControlFrame) {
+        _currentFrameOpcode = frame_header.opcode;
+        _currentFrameCount += 1;
+    }
+    
+    if (frame_header.payload_length == 0) {
+        if (isControlFrame) {
+            [self _handleFrameWithData:curData opCode:frame_header.opcode];
+        } else {
+            if (frame_header.fin) {
+                [self _handleFrameWithData:_currentFrameData opCode:frame_header.opcode];
+            } else {
+                // TODO add assert that opcode is not a control;
+                [self _readFrameContinue];
+            }
+        }
+    } else {
+        assert(frame_header.payload_length <= SIZE_T_MAX);
+        [self _addConsumerWithDataLength:(size_t)frame_header.payload_length callback:^(SRWebSocket *self, NSData *newData) {
+            if (isControlFrame) {
+                [self _handleFrameWithData:newData opCode:frame_header.opcode];
+            } else {
+                if (frame_header.fin) {
+                    [self _handleFrameWithData:self->_currentFrameData opCode:frame_header.opcode];
+                } else {
+                    // TODO add assert that opcode is not a control;
+                    [self _readFrameContinue];
+                }
+                
+            }
+        } readToCurrentFrame:!isControlFrame unmaskBytes:frame_header.masked];
+    }
+}
+
+/* From RFC:
+
+ 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
+ +-+-+-+-+-------+-+-------------+-------------------------------+
+ |F|R|R|R| opcode|M| Payload len |    Extended payload length    |
+ |I|S|S|S|  (4)  |A|     (7)     |             (16/64)           |
+ |N|V|V|V|       |S|             |   (if payload len==126/127)   |
+ | |1|2|3|       |K|             |                               |
+ +-+-+-+-+-------+-+-------------+ - - - - - - - - - - - - - - - +
+ |     Extended payload length continued, if payload len == 127  |
+ + - - - - - - - - - - - - - - - +-------------------------------+
+ |                               |Masking-key, if MASK set to 1  |
+ +-------------------------------+-------------------------------+
+ | Masking-key (continued)       |          Payload Data         |
+ +-------------------------------- - - - - - - - - - - - - - - - +
+ :                     Payload Data continued ...                :
+ + - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - +
+ |                     Payload Data continued ...                |
+ +---------------------------------------------------------------+
+ */
+
+static const uint8_t SRFinMask          = 0x80;
+static const uint8_t SROpCodeMask       = 0x0F;
+static const uint8_t SRRsvMask          = 0x70;
+static const uint8_t SRMaskMask         = 0x80;
+static const uint8_t SRPayloadLenMask   = 0x7F;
+
+
+- (void)_readFrameContinue;
+{
+    assert((_currentFrameCount == 0 && _currentFrameOpcode == 0) || (_currentFrameCount > 0 && _currentFrameOpcode > 0));
+
+    [self _addConsumerWithDataLength:2 callback:^(SRWebSocket *self, NSData *data) {
+        __block frame_header header = {0};
+        
+        const uint8_t *headerBuffer = data.bytes;
+        assert(data.length >= 2);
+        
+        if (headerBuffer[0] & SRRsvMask) {
+            [self _closeWithProtocolError:@"Server used RSV bits"];
+            return;
+        }
+        
+        uint8_t receivedOpcode = (SROpCodeMask & headerBuffer[0]);
+        
+        BOOL isControlFrame = (receivedOpcode == SROpCodePing || receivedOpcode == SROpCodePong || receivedOpcode == SROpCodeConnectionClose);
+        
+        if (!isControlFrame && receivedOpcode != 0 && self->_currentFrameCount > 0) {
+            [self _closeWithProtocolError:@"all data frames after the initial data frame must have opcode 0"];
+            return;
+        }
+        
+        if (receivedOpcode == 0 && self->_currentFrameCount == 0) {
+            [self _closeWithProtocolError:@"cannot continue a message"];
+            return;
+        }
+        
+        header.opcode = receivedOpcode == 0 ? self->_currentFrameOpcode : receivedOpcode;
+        
+        header.fin = !!(SRFinMask & headerBuffer[0]);
+        
+        
+        header.masked = !!(SRMaskMask & headerBuffer[1]);
+        header.payload_length = SRPayloadLenMask & headerBuffer[1];
+        
+        headerBuffer = NULL;
+        
+        if (header.masked) {
+            [self _closeWithProtocolError:@"Client must receive unmasked data"];
+        }
+
+        size_t extra_bytes_needed = header.masked ? sizeof(self->_currentReadMaskKey) : 0;
+
+        if (header.payload_length == 126) {
+            extra_bytes_needed += sizeof(uint16_t);
+        } else if (header.payload_length == 127) {
+            extra_bytes_needed += sizeof(uint64_t);
+        }
+        
+        if (extra_bytes_needed == 0) {
+            [self _handleFrameHeader:header curData:self->_currentFrameData];
+        } else {
+            [self _addConsumerWithDataLength:extra_bytes_needed callback:^(SRWebSocket *self, NSData *data) {
+                size_t mapped_size = data.length;
+                const void *mapped_buffer = data.bytes;
+                size_t offset = 0;
+                
+                if (header.payload_length == 126) {
+                    assert(mapped_size >= sizeof(uint16_t));
+                    uint16_t newLen = EndianU16_BtoN(*(uint16_t *)(mapped_buffer));
+                    header.payload_length = newLen;
+                    offset += sizeof(uint16_t);
+                } else if (header.payload_length == 127) {
+                    assert(mapped_size >= sizeof(uint64_t));
+                    header.payload_length = EndianU64_BtoN(*(uint64_t *)(mapped_buffer));
+                    offset += sizeof(uint64_t);
+                } else {
+                    assert(header.payload_length < 126 && header.payload_length >= 0);
+                }
+                
+                
+                if (header.masked) {
+                  assert(mapped_size >= sizeof(self->_currentReadMaskOffset) + offset);
+                  memcpy(self->_currentReadMaskKey,
+                         ((uint8_t *)mapped_buffer) + offset,
+                         sizeof(self->_currentReadMaskKey));
+                }
+                
+                [self _handleFrameHeader:header curData:self->_currentFrameData];
+            } readToCurrentFrame:NO unmaskBytes:NO];
+        }
+    } readToCurrentFrame:NO unmaskBytes:NO];
+}
+
+- (void)_readFrameNew;
+{
+  dispatch_async(_workQueue, ^{
+    [self->_currentFrameData setLength:0];
+
+    self->_currentFrameOpcode = 0;
+    self->_currentFrameCount = 0;
+    self->_readOpCount = 0;
+    self->_currentStringScanPosition = 0;
+
+    [self _readFrameContinue];
+  });
+}
+
+- (void)_pumpWriting;
+{
+    [self assertOnWorkQueue];
+    
+    NSUInteger dataLength = _outputBuffer.length;
+    if (dataLength - _outputBufferOffset > 0 && _outputStream.hasSpaceAvailable) {
+        NSInteger bytesWritten = [_outputStream write:_outputBuffer.bytes + _outputBufferOffset maxLength:dataLength - _outputBufferOffset];
+        if (bytesWritten == -1) {
+            [self _failWithError:[NSError errorWithDomain:SRWebSocketErrorDomain code:2145 userInfo:[NSDictionary dictionaryWithObject:@"Error writing to stream" forKey:NSLocalizedDescriptionKey]]];
+             return;
+        }
+        
+        _outputBufferOffset += bytesWritten;
+        
+        if (_outputBufferOffset > 4096 && _outputBufferOffset > (_outputBuffer.length >> 1)) {
+            _outputBuffer = [[NSMutableData alloc] initWithBytes:(char *)_outputBuffer.bytes + _outputBufferOffset length:_outputBuffer.length - _outputBufferOffset];
+            _outputBufferOffset = 0;
+        }
+    }
+    
+    if (_closeWhenFinishedWriting && 
+        _outputBuffer.length - _outputBufferOffset == 0 && 
+        (_inputStream.streamStatus != NSStreamStatusNotOpen &&
+         _inputStream.streamStatus != NSStreamStatusClosed) &&
+        !_sentClose) {
+        _sentClose = YES;
+            
+        [_outputStream close];
+        [_inputStream close];
+        
+        
+        for (NSArray *runLoop in [_scheduledRunloops copy]) {
+            [self unscheduleFromRunLoop:[runLoop objectAtIndex:0] forMode:[runLoop objectAtIndex:1]];
+        }
+        
+        if (!_failed) {
+            [self _performDelegateBlock:^{
+                if ([self.delegate respondsToSelector:@selector(webSocket:didCloseWithCode:reason:wasClean:)]) {
+                  [self.delegate webSocket:self
+                          didCloseWithCode:self->_closeCode
+                                    reason:self->_closeReason
+                                  wasClean:YES];
+                }
+            }];
+        }
+        
+        _selfRetain = nil;
+    }
+}
+
+- (void)_addConsumerWithScanner:(stream_scanner)consumer callback:(data_callback)callback;
+{
+    [self assertOnWorkQueue];
+    [self _addConsumerWithScanner:consumer callback:callback dataLength:0];
+}
+
+- (void)_addConsumerWithDataLength:(size_t)dataLength callback:(data_callback)callback readToCurrentFrame:(BOOL)readToCurrentFrame unmaskBytes:(BOOL)unmaskBytes;
+{   
+    [self assertOnWorkQueue];
+    assert(dataLength);
+    
+    [_consumers addObject:[_consumerPool consumerWithScanner:nil handler:callback bytesNeeded:dataLength readToCurrentFrame:readToCurrentFrame unmaskBytes:unmaskBytes]];
+    [self _pumpScanner];
+}
+
+- (void)_addConsumerWithScanner:(stream_scanner)consumer callback:(data_callback)callback dataLength:(size_t)dataLength;
+{    
+    [self assertOnWorkQueue];
+    [_consumers addObject:[_consumerPool consumerWithScanner:consumer handler:callback bytesNeeded:dataLength readToCurrentFrame:NO unmaskBytes:NO]];
+    [self _pumpScanner];
+}
+
+
+static const char CRLFCRLFBytes[] = {'\r', '\n', '\r', '\n'};
+
+- (void)_readUntilHeaderCompleteWithCallback:(data_callback)dataHandler;
+{
+    [self _readUntilBytes:CRLFCRLFBytes length:sizeof(CRLFCRLFBytes) callback:dataHandler];
+}
+
+- (void)_readUntilBytes:(const void *)bytes length:(size_t)length callback:(data_callback)dataHandler;
+{
+    // TODO optimize so this can continue from where we last searched
+    stream_scanner consumer = ^size_t(NSData *data) {
+        __block size_t found_size = 0;
+        __block size_t match_count = 0;
+        
+        size_t size = data.length;
+        const unsigned char *buffer = data.bytes;
+        for (size_t i = 0; i < size; i++ ) {
+            if (((const unsigned char *)buffer)[i] == ((const unsigned char *)bytes)[match_count]) {
+                match_count += 1;
+                if (match_count == length) {
+                    found_size = i + 1;
+                    break;
+                }
+            } else {
+                match_count = 0;
+            }
+        }
+        return found_size;
+    };
+    [self _addConsumerWithScanner:consumer callback:dataHandler];
+}
+
+
+// Returns true if did work
+- (BOOL)_innerPumpScanner {
+    
+    BOOL didWork = NO;
+    
+    if (self.readyState >= SR_CLOSING) {
+        return didWork;
+    }
+    
+    if (!_consumers.count) {
+        return didWork;
+    }
+    
+    size_t curSize = _readBuffer.length - _readBufferOffset;
+    if (!curSize) {
+        return didWork;
+    }
+    
+    SRIOConsumer *consumer = [_consumers objectAtIndex:0];
+    
+    size_t bytesNeeded = consumer.bytesNeeded;
+    
+    size_t foundSize = 0;
+    if (consumer.consumer) {
+        NSData *tempView = [NSData dataWithBytesNoCopy:(char *)_readBuffer.bytes + _readBufferOffset length:_readBuffer.length - _readBufferOffset freeWhenDone:NO];  
+        foundSize = consumer.consumer(tempView);
+    } else {
+        assert(consumer.bytesNeeded);
+        if (curSize >= bytesNeeded) {
+            foundSize = bytesNeeded;
+        } else if (consumer.readToCurrentFrame) {
+            foundSize = curSize;
+        }
+    }
+    
+    NSData *slice = nil;
+    if (consumer.readToCurrentFrame || foundSize) {
+        NSRange sliceRange = NSMakeRange(_readBufferOffset, foundSize);
+        slice = [_readBuffer subdataWithRange:sliceRange];
+        
+        _readBufferOffset += foundSize;
+        
+        if (_readBufferOffset > 4096 && _readBufferOffset > (_readBuffer.length >> 1)) {
+            _readBuffer = [[NSMutableData alloc] initWithBytes:(char *)_readBuffer.bytes + _readBufferOffset length:_readBuffer.length - _readBufferOffset];            _readBufferOffset = 0;
+        }
+        
+        if (consumer.unmaskBytes) {
+            NSMutableData *mutableSlice = [slice mutableCopy];
+            
+            NSUInteger len = mutableSlice.length;
+            uint8_t *bytes = mutableSlice.mutableBytes;
+            
+            for (NSUInteger i = 0; i < len; i++) {
+                bytes[i] = bytes[i] ^ _currentReadMaskKey[_currentReadMaskOffset % sizeof(_currentReadMaskKey)];
+                _currentReadMaskOffset += 1;
+            }
+            
+            slice = mutableSlice;
+        }
+        
+        if (consumer.readToCurrentFrame) {
+            [_currentFrameData appendData:slice];
+            
+            _readOpCount += 1;
+            
+            if (_currentFrameOpcode == SROpCodeTextFrame) {
+                // Validate UTF8 stuff.
+                size_t currentDataSize = _currentFrameData.length;
+                if (_currentFrameOpcode == SROpCodeTextFrame && currentDataSize > 0) {
+                    // TODO: Optimize the crap out of this.  Don't really have to copy all the data each time
+                    
+                    size_t scanSize = currentDataSize - _currentStringScanPosition;
+                    
+                    NSData *scan_data = [_currentFrameData subdataWithRange:NSMakeRange(_currentStringScanPosition, scanSize)];
+                    int32_t valid_utf8_size = validate_dispatch_data_partial_string(scan_data);
+                    
+                    if (valid_utf8_size == -1) {
+                        [self closeWithCode:SRStatusCodeInvalidUTF8 reason:@"Text frames must be valid UTF-8"];
+                        dispatch_async(_workQueue, ^{
+                            [self _disconnect];
+                        });
+                        return didWork;
+                    } else {
+                        _currentStringScanPosition += valid_utf8_size;
+                    }
+                } 
+                
+            }
+            
+            consumer.bytesNeeded -= foundSize;
+            
+            if (consumer.bytesNeeded == 0) {
+                [_consumers removeObjectAtIndex:0];
+                consumer.handler(self, nil);
+                [_consumerPool returnConsumer:consumer];
+                didWork = YES;
+            }
+        } else if (foundSize) {
+            [_consumers removeObjectAtIndex:0];
+            consumer.handler(self, slice);
+            [_consumerPool returnConsumer:consumer];
+            didWork = YES;
+        }
+    }
+    return didWork;
+}
+
+-(void)_pumpScanner;
+{
+    [self assertOnWorkQueue];
+    
+    if (!_isPumping) {
+        _isPumping = YES;
+    } else {
+        return;
+    }
+    
+    while ([self _innerPumpScanner]) {
+        
+    }
+    
+    _isPumping = NO;
+}
+
+//#define NOMASK
+
+static const size_t SRFrameHeaderOverhead = 32;
+
+- (void)_sendFrameWithOpcode:(SROpCode)opcode data:(id)data;
+{
+    [self assertOnWorkQueue];
+    
+    if (nil == data) {
+        return;
+    }
+    
+    NSAssert([data isKindOfClass:[NSData class]] || [data isKindOfClass:[NSString class]], @"NSString or NSData");
+    
+    size_t payloadLength = [data isKindOfClass:[NSString class]] ? [(NSString *)data lengthOfBytesUsingEncoding:NSUTF8StringEncoding] : [data length];
+        
+    NSMutableData *frame = [[NSMutableData alloc] initWithLength:payloadLength + SRFrameHeaderOverhead];
+    if (!frame) {
+        [self closeWithCode:SRStatusCodeMessageTooBig reason:@"Message too big"];
+        return;
+    }
+    uint8_t *frame_buffer = (uint8_t *)[frame mutableBytes];
+    
+    // set fin
+    frame_buffer[0] = SRFinMask | opcode;
+    
+    BOOL useMask = YES;
+#ifdef NOMASK
+    useMask = NO;
+#endif
+    
+    if (useMask) {
+    // set the mask and header
+        frame_buffer[1] |= SRMaskMask;
+    }
+    
+    size_t frame_buffer_size = 2;
+    
+    const uint8_t *unmasked_payload = NULL;
+    if ([data isKindOfClass:[NSData class]]) {
+        unmasked_payload = (uint8_t *)[data bytes];
+    } else if ([data isKindOfClass:[NSString class]]) {
+        unmasked_payload =  (const uint8_t *)[data UTF8String];
+    } else {
+        return;
+    }
+    
+    if (payloadLength < 126) {
+        frame_buffer[1] |= payloadLength;
+    } else if (payloadLength <= UINT16_MAX) {
+        frame_buffer[1] |= 126;
+        *((uint16_t *)(frame_buffer + frame_buffer_size)) = EndianU16_BtoN((uint16_t)payloadLength);
+        frame_buffer_size += sizeof(uint16_t);
+    } else {
+        frame_buffer[1] |= 127;
+        *((uint64_t *)(frame_buffer + frame_buffer_size)) = EndianU64_BtoN((uint64_t)payloadLength);
+        frame_buffer_size += sizeof(uint64_t);
+    }
+        
+    if (!useMask) {
+        for (size_t i = 0; i < payloadLength; i++) {
+            frame_buffer[frame_buffer_size] = unmasked_payload[i];
+            frame_buffer_size += 1;
+        }
+    } else {
+        uint8_t *mask_key = frame_buffer + frame_buffer_size;
+        BOOL success = !SecRandomCopyBytes(kSecRandomDefault, sizeof(uint32_t), (uint8_t *)mask_key);
+        assert(success);
+        frame_buffer_size += sizeof(uint32_t);
+        
+        // TODO: could probably optimize this with SIMD
+        for (size_t i = 0; i < payloadLength; i++) {
+            frame_buffer[frame_buffer_size] = unmasked_payload[i] ^ mask_key[i % sizeof(uint32_t)];
+            frame_buffer_size += 1;
+        }
+    }
+
+    assert(frame_buffer_size <= [frame length]);
+    frame.length = frame_buffer_size;
+    
+    [self _writeData:frame];
+}
+
+- (void)stream:(NSStream *)aStream handleEvent:(NSStreamEvent)eventCode;
+{
+    if (_secure && !_pinnedCertFound && (eventCode == NSStreamEventHasBytesAvailable || eventCode == NSStreamEventHasSpaceAvailable)) {
+        
+        NSArray *sslCerts = [_urlRequest SR_SSLPinnedCertificates];
+        if (sslCerts) {
+            SecTrustRef secTrust = (__bridge SecTrustRef)[aStream propertyForKey:(__bridge id)kCFStreamPropertySSLPeerTrust];
+            if (secTrust) {
+                NSInteger numCerts = SecTrustGetCertificateCount(secTrust);
+                for (NSInteger i = 0; i < numCerts && !_pinnedCertFound; i++) {
+                    SecCertificateRef cert = SecTrustGetCertificateAtIndex(secTrust, i);
+                    NSData *certData = CFBridgingRelease(SecCertificateCopyData(cert));
+                    
+                    for (id ref in sslCerts) {
+                        SecCertificateRef trustedCert = (__bridge SecCertificateRef)ref;
+                        NSData *trustedCertData = CFBridgingRelease(SecCertificateCopyData(trustedCert));
+                        
+                        if ([trustedCertData isEqualToData:certData]) {
+                            _pinnedCertFound = YES;
+                            break;
+                        }
+                    }
+                }
+            }
+            
+            if (!_pinnedCertFound) {
+                dispatch_async(_workQueue, ^{
+                    [self _failWithError:[NSError errorWithDomain:SRWebSocketErrorDomain code:23556 userInfo:[NSDictionary dictionaryWithObject:[NSString stringWithFormat:@"Invalid server cert"] forKey:NSLocalizedDescriptionKey]]];
+                });
+                return;
+            }
+        }
+    }
+
+    dispatch_async(_workQueue, ^{
+        switch (eventCode) {
+            case NSStreamEventOpenCompleted: {
+                SRFastLog(@"NSStreamEventOpenCompleted %@", aStream);
+                if (self.readyState >= SR_CLOSING) {
+                    return;
+                }
+                assert(self->_readBuffer);
+
+                if (self.readyState == SR_CONNECTING && aStream == self->_inputStream) {
+                  [self didConnect];
+                }
+                [self _pumpWriting];
+                [self _pumpScanner];
+                break;
+            }
+                
+            case NSStreamEventErrorOccurred: {
+                SRFastLog(@"NSStreamEventErrorOccurred %@ %@", aStream, [[aStream streamError] copy]);
+                /// TODO specify error better!
+                [self _failWithError:aStream.streamError];
+                self->_readBufferOffset = 0;
+                [self->_readBuffer setLength:0];
+                break;
+                
+            }
+                
+            case NSStreamEventEndEncountered: {
+                [self _pumpScanner];
+                SRFastLog(@"NSStreamEventEndEncountered %@", aStream);
+                if (aStream.streamError) {
+                    [self _failWithError:aStream.streamError];
+                } else {
+                    if (self.readyState != SR_CLOSED) {
+                        self.readyState = SR_CLOSED;
+                        self->_selfRetain = nil;
+                    }
+
+                    if (!self->_sentClose && !self->_failed) {
+                      self->_sentClose = YES;
+                      // If we get closed in this state it's probably not clean because we should be
+                      // sending this when we send messages
+                      [self
+                          _performDelegateBlock:^{
+                            if ([self.delegate respondsToSelector:@selector(webSocket:didCloseWithCode:reason:wasClean:)]) {
+                              [self.delegate webSocket:self
+                                      didCloseWithCode:SRStatusCodeGoingAway
+                                                reason:@"Stream end encountered"
+                                              wasClean:NO];
+                            }
+                          }];
+                    }
+                }
+                
+                break;
+            }
+                
+            case NSStreamEventHasBytesAvailable: {
+                SRFastLog(@"NSStreamEventHasBytesAvailable %@", aStream);
+                enum EnumType : int { bufferSize = 2048 };
+                uint8_t buffer[bufferSize];
+
+                while (self->_inputStream.hasBytesAvailable) {
+                  NSInteger bytes_read = [self->_inputStream read:buffer maxLength:bufferSize];
+
+                  if (bytes_read > 0) {
+                    [self->_readBuffer appendBytes:buffer length:bytes_read];
+                  } else if (bytes_read < 0) {
+                    [self _failWithError:self->_inputStream.streamError];
+                  }
+
+                  if (bytes_read != bufferSize) {
+                    break;
+                  }
+                };
+                [self _pumpScanner];
+                break;
+            }
+                
+            case NSStreamEventHasSpaceAvailable: {
+                SRFastLog(@"NSStreamEventHasSpaceAvailable %@", aStream);
+                [self _pumpWriting];
+                break;
+            }
+                
+            default:
+                SRFastLog(@"(default)  %@", aStream);
+                break;
+        }
+    });
+}
+
+@end
+
+
+@implementation SRIOConsumer
+
+@synthesize bytesNeeded = _bytesNeeded;
+@synthesize consumer = _scanner;
+@synthesize handler = _handler;
+@synthesize readToCurrentFrame = _readToCurrentFrame;
+@synthesize unmaskBytes = _unmaskBytes;
+
+- (void)setupWithScanner:(stream_scanner)scanner handler:(data_callback)handler bytesNeeded:(size_t)bytesNeeded readToCurrentFrame:(BOOL)readToCurrentFrame unmaskBytes:(BOOL)unmaskBytes;
+{
+    _scanner = [scanner copy];
+    _handler = [handler copy];
+    _bytesNeeded = bytesNeeded;
+    _readToCurrentFrame = readToCurrentFrame;
+    _unmaskBytes = unmaskBytes;
+    assert(_scanner || _bytesNeeded);
+}
+
+
+@end
+
+
+@implementation SRIOConsumerPool {
+    NSUInteger _poolSize;
+    NSMutableArray *_bufferedConsumers;
+}
+
+- (id)initWithBufferCapacity:(NSUInteger)poolSize;
+{
+    self = [super init];
+    if (self) {
+        _poolSize = poolSize;
+        _bufferedConsumers = [[NSMutableArray alloc] initWithCapacity:poolSize];
+    }
+    return self;
+}
+
+- (id)init
+{
+    return [self initWithBufferCapacity:8];
+}
+
+- (SRIOConsumer *)consumerWithScanner:(stream_scanner)scanner handler:(data_callback)handler bytesNeeded:(size_t)bytesNeeded readToCurrentFrame:(BOOL)readToCurrentFrame unmaskBytes:(BOOL)unmaskBytes;
+{
+    SRIOConsumer *consumer = nil;
+    if (_bufferedConsumers.count) {
+        consumer = [_bufferedConsumers lastObject];
+        [_bufferedConsumers removeLastObject];
+    } else {
+        consumer = [[SRIOConsumer alloc] init];
+    }
+    
+    [consumer setupWithScanner:scanner handler:handler bytesNeeded:bytesNeeded readToCurrentFrame:readToCurrentFrame unmaskBytes:unmaskBytes];
+    
+    return consumer;
+}
+
+- (void)returnConsumer:(SRIOConsumer *)consumer;
+{
+    if (_bufferedConsumers.count < _poolSize) {
+        [_bufferedConsumers addObject:consumer];
+    }
+}
+
+@end
+
+
+@implementation  NSURLRequest (CertificateAdditions)
+
+- (NSArray *)SR_SSLPinnedCertificates;
+{
+    return [NSURLProtocol propertyForKey:@"SR_SSLPinnedCertificates" inRequest:self];
+}
+
+@end
+
+@implementation  NSMutableURLRequest (CertificateAdditions)
+
+- (NSArray *)SR_SSLPinnedCertificates;
+{
+    return [NSURLProtocol propertyForKey:@"SR_SSLPinnedCertificates" inRequest:self];
+}
+
+- (void)setSR_SSLPinnedCertificates:(NSArray *)SR_SSLPinnedCertificates;
+{
+    [NSURLProtocol setProperty:SR_SSLPinnedCertificates forKey:@"SR_SSLPinnedCertificates" inRequest:self];
+}
+
+@end
+
+@implementation NSURL (SRWebSocket)
+
+- (NSString *)SR_origin;
+{
+    NSString *scheme = [self.scheme lowercaseString];
+        
+    if ([scheme isEqualToString:@"wss"]) {
+        scheme = @"https";
+    } else if ([scheme isEqualToString:@"ws"]) {
+        scheme = @"http";
+    }
+    
+    if (self.port) {
+        return [NSString stringWithFormat:@"%@://%@:%@/", scheme, self.host, self.port];
+    } else {
+        return [NSString stringWithFormat:@"%@://%@/", scheme, self.host];
+    }
+}
+
+@end
+
+//#define SR_ENABLE_LOG
+
+static inline void SRFastLog(NSString *format, ...)  {
+#ifdef SR_ENABLE_LOG
+    __block va_list arg_list;
+    va_start (arg_list, format);
+    
+    NSString *formattedString = [[NSString alloc] initWithFormat:format arguments:arg_list];
+    
+    va_end(arg_list);
+    
+    NSLog(@"[SR] %@", formattedString);
+#endif
+}
+
+
+#ifdef HAS_ICU
+
+static inline int32_t validate_dispatch_data_partial_string(NSData *data) {
+    if ([data length] > INT32_MAX) {
+        // INT32_MAX is the limit so long as this Framework is using 32 bit ints everywhere.
+        return -1;
+    }
+
+    int32_t size = (int32_t)[data length];
+
+    const void * contents = [data bytes];
+    const uint8_t *str = (const uint8_t *)contents;
+    
+    UChar32 codepoint = 1;
+    int32_t offset = 0;
+    int32_t lastOffset = 0;
+    while(offset < size && codepoint > 0)  {
+        lastOffset = offset;
+        U8_NEXT(str, offset, size, codepoint);
+    }
+    
+    if (codepoint == -1) {
+        // Check to see if the last byte is valid or whether it was just continuing
+        if (!U8_IS_LEAD(str[lastOffset]) || U8_COUNT_TRAIL_BYTES(str[lastOffset]) + lastOffset < (int32_t)size) {
+            
+            size = -1;
+        } else {
+            uint8_t leadByte = str[lastOffset];
+            U8_MASK_LEAD_BYTE(leadByte, U8_COUNT_TRAIL_BYTES(leadByte));
+            
+            for (int i = lastOffset + 1; i < offset; i++) {
+                if (U8_IS_SINGLE(str[i]) || U8_IS_LEAD(str[i]) || !U8_IS_TRAIL(str[i])) {
+                    size = -1;
+                }
+            }
+            
+            if (size != -1) {
+                size = lastOffset;
+            }
+        }
+    }
+    
+    if (size != -1 && ![[NSString alloc] initWithBytesNoCopy:(char *)[data bytes] length:size encoding:NSUTF8StringEncoding freeWhenDone:NO]) {
+        size = -1;
+    }
+    
+    return size;
+}
+
+#else
+
+// This is a hack, and probably not optimal
+static inline int32_t validate_dispatch_data_partial_string(NSData *data) {
+    static const int maxCodepointSize = 3;
+    
+    for (int i = 0; i < maxCodepointSize; i++) {
+        NSString *str = [[NSString alloc] initWithBytesNoCopy:(char *)data.bytes length:data.length - i encoding:NSUTF8StringEncoding freeWhenDone:NO];
+        if (str) {
+            return data.length - i;
+        }
+    }
+    
+    return -1;
+}
+
+#endif
+
+static _SRRunLoopThread *networkThread = nil;
+static NSRunLoop *networkRunLoop = nil;
+
+@implementation NSRunLoop (SRWebSocket)
+
++ (NSRunLoop *)SR_networkRunLoop {
+    static dispatch_once_t onceToken;
+    dispatch_once(&onceToken, ^{
+        networkThread = [[_SRRunLoopThread alloc] init];
+        networkThread.name = @"com.squareup.SocketRocket.NetworkThread";
+        [networkThread start];
+        networkRunLoop = networkThread.runLoop;
+    });
+    
+    return networkRunLoop;
+}
+
+@end
+
+
+@implementation _SRRunLoopThread {
+    dispatch_group_t _waitGroup;
+}
+
+@synthesize runLoop = _runLoop;
+
+- (void)dealloc
+{
+    sr_dispatch_release(_waitGroup);
+}
+
+- (id)init
+{
+    self = [super init];
+    if (self) {
+        _waitGroup = dispatch_group_create();
+        dispatch_group_enter(_waitGroup);
+    }
+    return self;
+}
+
+- (void)main;
+{
+    @autoreleasepool {
+        _runLoop = [NSRunLoop currentRunLoop];
+        dispatch_group_leave(_waitGroup);
+        
+        NSTimer *timer = [[NSTimer alloc] initWithFireDate:[NSDate distantFuture] interval:0.0 target:nil selector:nil userInfo:nil repeats:NO];
+        [_runLoop addTimer:timer forMode:NSDefaultRunLoopMode];
+        
+        while ([_runLoop runMode:NSDefaultRunLoopMode beforeDate:[NSDate distantFuture]]) {
+            
+        }
+        assert(NO);
+    }
+}
+
+- (NSRunLoop *)runLoop;
+{
+    dispatch_group_wait(_waitGroup, DISPATCH_TIME_FOREVER);
+    return _runLoop;
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objc/Icon-120.png b/third_party/libwebrtc/examples/objc/Icon-120.png
new file mode 100644
index 0000000000..938fef477b
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/Icon-120.png
diff --git a/third_party/libwebrtc/examples/objc/Icon-180.png b/third_party/libwebrtc/examples/objc/Icon-180.png
new file mode 100644
index 0000000000..a5b7609680
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/Icon-180.png
diff --git a/third_party/libwebrtc/examples/objc/Icon.png b/third_party/libwebrtc/examples/objc/Icon.png
new file mode 100644
index 0000000000..55773ca9d9
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/Icon.png
diff --git a/third_party/libwebrtc/examples/objc/README b/third_party/libwebrtc/examples/objc/README
new file mode 100644
index 0000000000..bfe18b37c5
--- /dev/null
+++ b/third_party/libwebrtc/examples/objc/README
@@ -0,0 +1,3 @@
+This directory contains sample iOS and mac clients for http://apprtc.appspot.com
+
+See ../../app/webrtc/objc/README for information on how to use it.
diff --git a/third_party/libwebrtc/examples/objcnativeapi/Info.plist b/third_party/libwebrtc/examples/objcnativeapi/Info.plist
new file mode 100644
index 0000000000..cbc9e5f9f3
--- /dev/null
+++ b/third_party/libwebrtc/examples/objcnativeapi/Info.plist
@@ -0,0 +1,45 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+  <key>CFBundleDevelopmentRegion</key>
+  <string>en</string>
+  <key>CFBundleExecutable</key>
+  <string>$(EXECUTABLE_NAME)</string>
+  <key>CFBundleIdentifier</key>
+  <string>com.google.ObjCNativeAPIDemo</string>
+  <key>CFBundleInfoDictionaryVersion</key>
+  <string>6.0</string>
+  <key>CFBundleName</key>
+  <string>ObjCNativeAPIDemo</string>
+  <key>CFBundlePackageType</key>
+  <string>APPL</string>
+  <key>CFBundleShortVersionString</key>
+  <string>1.0</string>
+  <key>CFBundleVersion</key>
+  <string>1</string>
+  <key>LSRequiresIPhoneOS</key>
+  <true/>
+  <key>UIRequiredDeviceCapabilities</key>
+  <array>
+    <string>armv7</string>
+  </array>
+  <key>UISupportedInterfaceOrientations</key>
+  <array>
+    <string>UIInterfaceOrientationPortrait</string>
+    <string>UIInterfaceOrientationLandscapeLeft</string>
+    <string>UIInterfaceOrientationLandscapeRight</string>
+  </array>
+  <key>UISupportedInterfaceOrientations~ipad</key>
+  <array>
+    <string>UIInterfaceOrientationPortrait</string>
+    <string>UIInterfaceOrientationPortraitUpsideDown</string>
+    <string>UIInterfaceOrientationLandscapeLeft</string>
+    <string>UIInterfaceOrientationLandscapeRight</string>
+  </array>
+  <key>NSCameraUsageDescription</key>
+  <string>Camera access needed for video calling</string>
+  <key>NSMicrophoneUsageDescription</key>
+  <string>Microphone access needed for video calling</string>
+</dict>
+</plist>
diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/NADAppDelegate.h b/third_party/libwebrtc/examples/objcnativeapi/objc/NADAppDelegate.h
new file mode 100644
index 0000000000..02372dbfd2
--- /dev/null
+++ b/third_party/libwebrtc/examples/objcnativeapi/objc/NADAppDelegate.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+@interface NADAppDelegate : UIResponder <UIApplicationDelegate>
+
+@property(strong, nonatomic) UIWindow* window;
+
+@end
diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/NADAppDelegate.m b/third_party/libwebrtc/examples/objcnativeapi/objc/NADAppDelegate.m
new file mode 100644
index 0000000000..254dd3be76
--- /dev/null
+++ b/third_party/libwebrtc/examples/objcnativeapi/objc/NADAppDelegate.m
@@ -0,0 +1,63 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "NADAppDelegate.h"
+
+#import "NADViewController.h"
+
+@interface NADAppDelegate ()
+@end
+
+@implementation NADAppDelegate
+
+@synthesize window = _window;
+
+- (BOOL)application:(UIApplication *)application
+    didFinishLaunchingWithOptions:(NSDictionary *)launchOptions {
+  _window = [[UIWindow alloc] initWithFrame:[[UIScreen mainScreen] bounds]];
+  [_window makeKeyAndVisible];
+
+  NADViewController *viewController = [[NADViewController alloc] init];
+  _window.rootViewController = viewController;
+
+  return YES;
+}
+
+- (void)applicationWillResignActive:(UIApplication *)application {
+  // Sent when the application is about to move from active to inactive state. This can occur for
+  // certain types of temporary interruptions (such as an incoming phone call or SMS message) or
+  // when the user quits the application and it begins the transition to the background state. Use
+  // this method to pause ongoing tasks, disable timers, and invalidate graphics rendering
+  // callbacks. Games should use this method to pause the game.
+}
+
+- (void)applicationDidEnterBackground:(UIApplication *)application {
+  // Use this method to release shared resources, save user data, invalidate timers, and store
+  // enough application state information to restore your application to its current state in case
+  // it is terminated later. If your application supports background execution, this method is
+  // called instead of applicationWillTerminate: when the user quits.
+}
+
+- (void)applicationWillEnterForeground:(UIApplication *)application {
+  // Called as part of the transition from the background to the active state; here you can undo
+  // many of the changes made on entering the background.
+}
+
+- (void)applicationDidBecomeActive:(UIApplication *)application {
+  // Restart any tasks that were paused (or not yet started) while the application was inactive. If
+  // the application was previously in the background, optionally refresh the user interface.
+}
+
+- (void)applicationWillTerminate:(UIApplication *)application {
+  // Called when the application is about to terminate. Save data if appropriate. See also
+  // applicationDidEnterBackground:.
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/NADViewController.h b/third_party/libwebrtc/examples/objcnativeapi/objc/NADViewController.h
new file mode 100644
index 0000000000..c43bebb52d
--- /dev/null
+++ b/third_party/libwebrtc/examples/objcnativeapi/objc/NADViewController.h
@@ -0,0 +1,15 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+@interface NADViewController : UIViewController
+
+@end
diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/NADViewController.mm b/third_party/libwebrtc/examples/objcnativeapi/objc/NADViewController.mm
new file mode 100644
index 0000000000..fd244799f8
--- /dev/null
+++ b/third_party/libwebrtc/examples/objcnativeapi/objc/NADViewController.mm
@@ -0,0 +1,154 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "NADViewController.h"
+
+#import "sdk/objc/base/RTCVideoRenderer.h"
+#import "sdk/objc/components/capturer/RTCCameraVideoCapturer.h"
+#import "sdk/objc/components/renderer/metal/RTCMTLVideoView.h"
+#import "sdk/objc/helpers/RTCCameraPreviewView.h"
+
+#include <memory>
+
+#include "examples/objcnativeapi/objc/objc_call_client.h"
+
+@interface NADViewController ()
+
+@property(nonatomic) RTC_OBJC_TYPE(RTCCameraVideoCapturer) * capturer;
+@property(nonatomic) RTC_OBJC_TYPE(RTCCameraPreviewView) * localVideoView;
+@property(nonatomic) __kindof UIView<RTC_OBJC_TYPE(RTCVideoRenderer)> *remoteVideoView;
+@property(nonatomic) UIButton *callButton;
+@property(nonatomic) UIButton *hangUpButton;
+
+@end
+
+@implementation NADViewController {
+  std::unique_ptr<webrtc_examples::ObjCCallClient> _call_client;
+
+  UIView *_view;
+}
+
+@synthesize capturer = _capturer;
+@synthesize localVideoView = _localVideoView;
+@synthesize remoteVideoView = _remoteVideoView;
+@synthesize callButton = _callButton;
+@synthesize hangUpButton = _hangUpButton;
+
+#pragma mark - View controller lifecycle
+
+- (void)loadView {
+  _view = [[UIView alloc] initWithFrame:CGRectZero];
+
+  _remoteVideoView = [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectZero];
+  _remoteVideoView.translatesAutoresizingMaskIntoConstraints = NO;
+  [_view addSubview:_remoteVideoView];
+
+  _localVideoView = [[RTC_OBJC_TYPE(RTCCameraPreviewView) alloc] initWithFrame:CGRectZero];
+  _localVideoView.translatesAutoresizingMaskIntoConstraints = NO;
+  [_view addSubview:_localVideoView];
+
+  _callButton = [UIButton buttonWithType:UIButtonTypeSystem];
+  _callButton.translatesAutoresizingMaskIntoConstraints = NO;
+  [_callButton setTitle:@"Call" forState:UIControlStateNormal];
+  [_callButton addTarget:self action:@selector(call:) forControlEvents:UIControlEventTouchUpInside];
+  [_view addSubview:_callButton];
+
+  _hangUpButton = [UIButton buttonWithType:UIButtonTypeSystem];
+  _hangUpButton.translatesAutoresizingMaskIntoConstraints = NO;
+  [_hangUpButton setTitle:@"Hang up" forState:UIControlStateNormal];
+  [_hangUpButton addTarget:self
+                    action:@selector(hangUp:)
+          forControlEvents:UIControlEventTouchUpInside];
+  [_view addSubview:_hangUpButton];
+
+  UILayoutGuide *margin = _view.layoutMarginsGuide;
+  [_remoteVideoView.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor].active = YES;
+  [_remoteVideoView.topAnchor constraintEqualToAnchor:margin.topAnchor].active = YES;
+  [_remoteVideoView.trailingAnchor constraintEqualToAnchor:margin.trailingAnchor].active = YES;
+  [_remoteVideoView.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor].active = YES;
+
+  [_localVideoView.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor constant:8.0].active =
+      YES;
+  [_localVideoView.topAnchor constraintEqualToAnchor:margin.topAnchor constant:8.0].active = YES;
+  [_localVideoView.widthAnchor constraintEqualToConstant:60].active = YES;
+  [_localVideoView.heightAnchor constraintEqualToConstant:60].active = YES;
+
+  [_callButton.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor constant:8.0].active =
+      YES;
+  [_callButton.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor constant:8.0].active = YES;
+  [_callButton.widthAnchor constraintEqualToConstant:100].active = YES;
+  [_callButton.heightAnchor constraintEqualToConstant:40].active = YES;
+
+  [_hangUpButton.trailingAnchor constraintEqualToAnchor:margin.trailingAnchor constant:8.0].active =
+      YES;
+  [_hangUpButton.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor constant:8.0].active =
+      YES;
+  [_hangUpButton.widthAnchor constraintEqualToConstant:100].active = YES;
+  [_hangUpButton.heightAnchor constraintEqualToConstant:40].active = YES;
+
+  self.view = _view;
+}
+
+- (void)viewDidLoad {
+  [super viewDidLoad];
+
+  self.capturer = [[RTC_OBJC_TYPE(RTCCameraVideoCapturer) alloc] init];
+  self.localVideoView.captureSession = self.capturer.captureSession;
+
+  _call_client.reset(new webrtc_examples::ObjCCallClient());
+
+  // Start capturer.
+  AVCaptureDevice *selectedDevice = nil;
+  NSArray<AVCaptureDevice *> *captureDevices =
+      [RTC_OBJC_TYPE(RTCCameraVideoCapturer) captureDevices];
+  for (AVCaptureDevice *device in captureDevices) {
+    if (device.position == AVCaptureDevicePositionFront) {
+      selectedDevice = device;
+      break;
+    }
+  }
+
+  AVCaptureDeviceFormat *selectedFormat = nil;
+  int targetWidth = 640;
+  int targetHeight = 480;
+  int currentDiff = INT_MAX;
+  NSArray<AVCaptureDeviceFormat *> *formats =
+      [RTC_OBJC_TYPE(RTCCameraVideoCapturer) supportedFormatsForDevice:selectedDevice];
+  for (AVCaptureDeviceFormat *format in formats) {
+    CMVideoDimensions dimension = CMVideoFormatDescriptionGetDimensions(format.formatDescription);
+    FourCharCode pixelFormat = CMFormatDescriptionGetMediaSubType(format.formatDescription);
+    int diff = abs(targetWidth - dimension.width) + abs(targetHeight - dimension.height);
+    if (diff < currentDiff) {
+      selectedFormat = format;
+      currentDiff = diff;
+    } else if (diff == currentDiff && pixelFormat == [_capturer preferredOutputPixelFormat]) {
+      selectedFormat = format;
+    }
+  }
+
+  [self.capturer startCaptureWithDevice:selectedDevice format:selectedFormat fps:30];
+}
+
+- (void)didReceiveMemoryWarning {
+  [super didReceiveMemoryWarning];
+  // Dispose of any resources that can be recreated.
+}
+
+#pragma mark - Actions
+
+- (IBAction)call:(id)sender {
+  _call_client->Call(self.capturer, self.remoteVideoView);
+}
+
+- (IBAction)hangUp:(id)sender {
+  _call_client->Hangup();
+}
+
+@end
diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/main.m b/third_party/libwebrtc/examples/objcnativeapi/objc/main.m
new file mode 100644
index 0000000000..2c3b5fbbfb
--- /dev/null
+++ b/third_party/libwebrtc/examples/objcnativeapi/objc/main.m
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+#import "NADAppDelegate.h"
+
+int main(int argc, char* argv[]) {
+  @autoreleasepool {
+    return UIApplicationMain(argc, argv, nil, NSStringFromClass([NADAppDelegate class]));
+  }
+}
diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.h b/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.h
new file mode 100644
index 0000000000..cb8501d9ce
--- /dev/null
+++ b/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_OBJCNATIVEAPI_OBJCCALLCLIENT_H_
+#define EXAMPLES_OBJCNATIVEAPI_OBJCCALLCLIENT_H_
+
+#include <memory>
+#include <string>
+
+#import "sdk/objc/base/RTCMacros.h"
+
+#include "api/peer_connection_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "rtc_base/synchronization/mutex.h"
+
+@class RTC_OBJC_TYPE(RTCVideoCapturer);
+@protocol RTC_OBJC_TYPE
+(RTCVideoRenderer);
+
+namespace webrtc_examples {
+
+class ObjCCallClient {
+ public:
+  ObjCCallClient();
+
+  void Call(RTC_OBJC_TYPE(RTCVideoCapturer) * capturer,
+            id<RTC_OBJC_TYPE(RTCVideoRenderer)> remote_renderer);
+  void Hangup();
+
+ private:
+  class PCObserver : public webrtc::PeerConnectionObserver {
+   public:
+    explicit PCObserver(ObjCCallClient* client);
+
+    void OnSignalingChange(webrtc::PeerConnectionInterface::SignalingState new_state) override;
+    void OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;
+    void OnRenegotiationNeeded() override;
+    void OnIceConnectionChange(
+        webrtc::PeerConnectionInterface::IceConnectionState new_state) override;
+    void OnIceGatheringChange(
+        webrtc::PeerConnectionInterface::IceGatheringState new_state) override;
+    void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
+
+   private:
+    ObjCCallClient* const client_;
+  };
+
+  void CreatePeerConnectionFactory() RTC_RUN_ON(thread_checker_);
+  void CreatePeerConnection() RTC_RUN_ON(thread_checker_);
+  void Connect() RTC_RUN_ON(thread_checker_);
+
+  webrtc::SequenceChecker thread_checker_;
+
+  bool call_started_ RTC_GUARDED_BY(thread_checker_);
+
+  const std::unique_ptr<PCObserver> pc_observer_;
+
+  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf_ RTC_GUARDED_BY(thread_checker_);
+  std::unique_ptr<rtc::Thread> network_thread_ RTC_GUARDED_BY(thread_checker_);
+  std::unique_ptr<rtc::Thread> worker_thread_ RTC_GUARDED_BY(thread_checker_);
+  std::unique_ptr<rtc::Thread> signaling_thread_ RTC_GUARDED_BY(thread_checker_);
+
+  std::unique_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> remote_sink_
+      RTC_GUARDED_BY(thread_checker_);
+  rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> video_source_
+      RTC_GUARDED_BY(thread_checker_);
+
+  webrtc::Mutex pc_mutex_;
+  rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_ RTC_GUARDED_BY(pc_mutex_);
+};
+
+}  // namespace webrtc_examples
+
+#endif  // EXAMPLES_OBJCNATIVEAPI_OBJCCALLCLIENT_H_
diff --git a/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.mm b/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.mm
new file mode 100644
index 0000000000..7dd57b499b
--- /dev/null
+++ b/third_party/libwebrtc/examples/objcnativeapi/objc/objc_call_client.mm
@@ -0,0 +1,238 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/objcnativeapi/objc/objc_call_client.h"
+
+#include <memory>
+#include <utility>
+
+#import "sdk/objc/base/RTCVideoRenderer.h"
+#import "sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h"
+#import "sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h"
+#import "sdk/objc/helpers/RTCCameraPreviewView.h"
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "media/engine/webrtc_media_engine.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "sdk/objc/native/api/video_capturer.h"
+#include "sdk/objc/native/api/video_decoder_factory.h"
+#include "sdk/objc/native/api/video_encoder_factory.h"
+#include "sdk/objc/native/api/video_renderer.h"
+
+namespace webrtc_examples {
+
+namespace {
+
+class CreateOfferObserver : public webrtc::CreateSessionDescriptionObserver {
+ public:
+  explicit CreateOfferObserver(rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc);
+
+  void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
+  void OnFailure(webrtc::RTCError error) override;
+
+ private:
+  const rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_;
+};
+
+class SetRemoteSessionDescriptionObserver : public webrtc::SetRemoteDescriptionObserverInterface {
+ public:
+  void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override;
+};
+
+class SetLocalSessionDescriptionObserver : public webrtc::SetLocalDescriptionObserverInterface {
+ public:
+  void OnSetLocalDescriptionComplete(webrtc::RTCError error) override;
+};
+
+}  // namespace
+
+ObjCCallClient::ObjCCallClient()
+    : call_started_(false), pc_observer_(std::make_unique<PCObserver>(this)) {
+  thread_checker_.Detach();
+  CreatePeerConnectionFactory();
+}
+
+void ObjCCallClient::Call(RTC_OBJC_TYPE(RTCVideoCapturer) * capturer,
+                          id<RTC_OBJC_TYPE(RTCVideoRenderer)> remote_renderer) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  webrtc::MutexLock lock(&pc_mutex_);
+  if (call_started_) {
+    RTC_LOG(LS_WARNING) << "Call already started.";
+    return;
+  }
+  call_started_ = true;
+
+  remote_sink_ = webrtc::ObjCToNativeVideoRenderer(remote_renderer);
+
+  video_source_ =
+      webrtc::ObjCToNativeVideoCapturer(capturer, signaling_thread_.get(), worker_thread_.get());
+
+  CreatePeerConnection();
+  Connect();
+}
+
+void ObjCCallClient::Hangup() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  call_started_ = false;
+
+  {
+    webrtc::MutexLock lock(&pc_mutex_);
+    if (pc_ != nullptr) {
+      pc_->Close();
+      pc_ = nullptr;
+    }
+  }
+
+  remote_sink_ = nullptr;
+  video_source_ = nullptr;
+}
+
+void ObjCCallClient::CreatePeerConnectionFactory() {
+  network_thread_ = rtc::Thread::CreateWithSocketServer();
+  network_thread_->SetName("network_thread", nullptr);
+  RTC_CHECK(network_thread_->Start()) << "Failed to start thread";
+
+  worker_thread_ = rtc::Thread::Create();
+  worker_thread_->SetName("worker_thread", nullptr);
+  RTC_CHECK(worker_thread_->Start()) << "Failed to start thread";
+
+  signaling_thread_ = rtc::Thread::Create();
+  signaling_thread_->SetName("signaling_thread", nullptr);
+  RTC_CHECK(signaling_thread_->Start()) << "Failed to start thread";
+
+  webrtc::PeerConnectionFactoryDependencies dependencies;
+  dependencies.network_thread = network_thread_.get();
+  dependencies.worker_thread = worker_thread_.get();
+  dependencies.signaling_thread = signaling_thread_.get();
+  dependencies.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
+  cricket::MediaEngineDependencies media_deps;
+  media_deps.task_queue_factory = dependencies.task_queue_factory.get();
+  media_deps.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
+  media_deps.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
+  media_deps.video_encoder_factory = webrtc::ObjCToNativeVideoEncoderFactory(
+      [[RTC_OBJC_TYPE(RTCDefaultVideoEncoderFactory) alloc] init]);
+  media_deps.video_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory(
+      [[RTC_OBJC_TYPE(RTCDefaultVideoDecoderFactory) alloc] init]);
+  media_deps.audio_processing = webrtc::AudioProcessingBuilder().Create();
+  dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
+  RTC_LOG(LS_INFO) << "Media engine created: " << dependencies.media_engine.get();
+  dependencies.call_factory = webrtc::CreateCallFactory();
+  dependencies.event_log_factory =
+      std::make_unique<webrtc::RtcEventLogFactory>(dependencies.task_queue_factory.get());
+  pcf_ = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
+  RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_.get();
+}
+
+void ObjCCallClient::CreatePeerConnection() {
+  webrtc::MutexLock lock(&pc_mutex_);
+  webrtc::PeerConnectionInterface::RTCConfiguration config;
+  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+  // Encryption has to be disabled for loopback to work.
+  webrtc::PeerConnectionFactoryInterface::Options options;
+  options.disable_encryption = true;
+  pcf_->SetOptions(options);
+  webrtc::PeerConnectionDependencies pc_dependencies(pc_observer_.get());
+  pc_ = pcf_->CreatePeerConnectionOrError(config, std::move(pc_dependencies)).MoveValue();
+  RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_.get();
+
+  rtc::scoped_refptr<webrtc::VideoTrackInterface> local_video_track =
+      pcf_->CreateVideoTrack("video", video_source_.get());
+  pc_->AddTransceiver(local_video_track);
+  RTC_LOG(LS_INFO) << "Local video sink set up: " << local_video_track.get();
+
+  for (const rtc::scoped_refptr<webrtc::RtpTransceiverInterface>& tranceiver :
+       pc_->GetTransceivers()) {
+    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track = tranceiver->receiver()->track();
+    if (track && track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) {
+      static_cast<webrtc::VideoTrackInterface*>(track.get())
+          ->AddOrUpdateSink(remote_sink_.get(), rtc::VideoSinkWants());
+      RTC_LOG(LS_INFO) << "Remote video sink set up: " << track.get();
+      break;
+    }
+  }
+}
+
+void ObjCCallClient::Connect() {
+  webrtc::MutexLock lock(&pc_mutex_);
+  pc_->CreateOffer(rtc::make_ref_counted<CreateOfferObserver>(pc_).get(),
+                   webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
+}
+
+ObjCCallClient::PCObserver::PCObserver(ObjCCallClient* client) : client_(client) {}
+
+void ObjCCallClient::PCObserver::OnSignalingChange(
+    webrtc::PeerConnectionInterface::SignalingState new_state) {
+  RTC_LOG(LS_INFO) << "OnSignalingChange: " << new_state;
+}
+
+void ObjCCallClient::PCObserver::OnDataChannel(
+    rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) {
+  RTC_LOG(LS_INFO) << "OnDataChannel";
+}
+
+void ObjCCallClient::PCObserver::OnRenegotiationNeeded() {
+  RTC_LOG(LS_INFO) << "OnRenegotiationNeeded";
+}
+
+void ObjCCallClient::PCObserver::OnIceConnectionChange(
+    webrtc::PeerConnectionInterface::IceConnectionState new_state) {
+  RTC_LOG(LS_INFO) << "OnIceConnectionChange: " << new_state;
+}
+
+void ObjCCallClient::PCObserver::OnIceGatheringChange(
+    webrtc::PeerConnectionInterface::IceGatheringState new_state) {
+  RTC_LOG(LS_INFO) << "OnIceGatheringChange: " << new_state;
+}
+
+void ObjCCallClient::PCObserver::OnIceCandidate(const webrtc::IceCandidateInterface* candidate) {
+  RTC_LOG(LS_INFO) << "OnIceCandidate: " << candidate->server_url();
+  webrtc::MutexLock lock(&client_->pc_mutex_);
+  RTC_DCHECK(client_->pc_ != nullptr);
+  client_->pc_->AddIceCandidate(candidate);
+}
+
+CreateOfferObserver::CreateOfferObserver(rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc)
+    : pc_(pc) {}
+
+void CreateOfferObserver::OnSuccess(webrtc::SessionDescriptionInterface* desc) {
+  std::string sdp;
+  desc->ToString(&sdp);
+  RTC_LOG(LS_INFO) << "Created offer: " << sdp;
+
+  // Ownership of desc was transferred to us, now we transfer it forward.
+  pc_->SetLocalDescription(absl::WrapUnique(desc),
+                           rtc::make_ref_counted<SetLocalSessionDescriptionObserver>());
+
+  // Generate a fake answer.
+  std::unique_ptr<webrtc::SessionDescriptionInterface> answer(
+      webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp));
+  pc_->SetRemoteDescription(std::move(answer),
+                            rtc::make_ref_counted<SetRemoteSessionDescriptionObserver>());
+}
+
+void CreateOfferObserver::OnFailure(webrtc::RTCError error) {
+  RTC_LOG(LS_INFO) << "Failed to create offer: " << error.message();
+}
+
+void SetRemoteSessionDescriptionObserver::OnSetRemoteDescriptionComplete(webrtc::RTCError error) {
+  RTC_LOG(LS_INFO) << "Set remote description: " << error.message();
+}
+
+void SetLocalSessionDescriptionObserver::OnSetLocalDescriptionComplete(webrtc::RTCError error) {
+  RTC_LOG(LS_INFO) << "Set local description: " << error.message();
+}
+
+}  // namespace webrtc_examples
diff --git a/third_party/libwebrtc/examples/peerconnection/OWNERS b/third_party/libwebrtc/examples/peerconnection/OWNERS
new file mode 100644
index 0000000000..0fba125734
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/OWNERS
@@ -0,0 +1 @@
+tommi@webrtc.org
diff --git a/third_party/libwebrtc/examples/peerconnection/client/conductor.cc b/third_party/libwebrtc/examples/peerconnection/client/conductor.cc
new file mode 100644
index 0000000000..965525abff
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/client/conductor.cc
@@ -0,0 +1,599 @@
+/*
+ *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/peerconnection/client/conductor.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/types/optional.h"
+#include "api/audio/audio_mixer.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/audio_options.h"
+#include "api/create_peerconnection_factory.h"
+#include "api/rtp_sender_interface.h"
+#include "api/video_codecs/builtin_video_decoder_factory.h"
+#include "api/video_codecs/builtin_video_encoder_factory.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "examples/peerconnection/client/defaults.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/video_capture/video_capture.h"
+#include "modules/video_capture/video_capture_factory.h"
+#include "p2p/base/port_allocator.h"
+#include "pc/video_track_source.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/rtc_certificate_generator.h"
+#include "rtc_base/strings/json.h"
+#include "test/vcm_capturer.h"
+
+namespace {
+// Names used for a IceCandidate JSON object.
+const char kCandidateSdpMidName[] = "sdpMid";
+const char kCandidateSdpMlineIndexName[] = "sdpMLineIndex";
+const char kCandidateSdpName[] = "candidate";
+
+// Names used for a SessionDescription JSON object.
+const char kSessionDescriptionTypeName[] = "type";
+const char kSessionDescriptionSdpName[] = "sdp";
+
+class DummySetSessionDescriptionObserver
+    : public webrtc::SetSessionDescriptionObserver {
+ public:
+  static rtc::scoped_refptr<DummySetSessionDescriptionObserver> Create() {
+    return rtc::make_ref_counted<DummySetSessionDescriptionObserver>();
+  }
+  virtual void OnSuccess() { RTC_LOG(LS_INFO) << __FUNCTION__; }
+  virtual void OnFailure(webrtc::RTCError error) {
+    RTC_LOG(LS_INFO) << __FUNCTION__ << " " << ToString(error.type()) << ": "
+                     << error.message();
+  }
+};
+
+class CapturerTrackSource : public webrtc::VideoTrackSource {
+ public:
+  static rtc::scoped_refptr<CapturerTrackSource> Create() {
+    const size_t kWidth = 640;
+    const size_t kHeight = 480;
+    const size_t kFps = 30;
+    std::unique_ptr<webrtc::test::VcmCapturer> capturer;
+    std::unique_ptr<webrtc::VideoCaptureModule::DeviceInfo> info(
+        webrtc::VideoCaptureFactory::CreateDeviceInfo());
+    if (!info) {
+      return nullptr;
+    }
+    int num_devices = info->NumberOfDevices();
+    for (int i = 0; i < num_devices; ++i) {
+      capturer = absl::WrapUnique(
+          webrtc::test::VcmCapturer::Create(kWidth, kHeight, kFps, i));
+      if (capturer) {
+        return rtc::make_ref_counted<CapturerTrackSource>(std::move(capturer));
+      }
+    }
+
+    return nullptr;
+  }
+
+ protected:
+  explicit CapturerTrackSource(
+      std::unique_ptr<webrtc::test::VcmCapturer> capturer)
+      : VideoTrackSource(/*remote=*/false), capturer_(std::move(capturer)) {}
+
+ private:
+  rtc::VideoSourceInterface<webrtc::VideoFrame>* source() override {
+    return capturer_.get();
+  }
+  std::unique_ptr<webrtc::test::VcmCapturer> capturer_;
+};
+
+}  // namespace
+
+Conductor::Conductor(PeerConnectionClient* client, MainWindow* main_wnd)
+    : peer_id_(-1), loopback_(false), client_(client), main_wnd_(main_wnd) {
+  client_->RegisterObserver(this);
+  main_wnd->RegisterObserver(this);
+}
+
+Conductor::~Conductor() {
+  RTC_DCHECK(!peer_connection_);
+}
+
+bool Conductor::connection_active() const {
+  return peer_connection_ != nullptr;
+}
+
+void Conductor::Close() {
+  client_->SignOut();
+  DeletePeerConnection();
+}
+
+bool Conductor::InitializePeerConnection() {
+  RTC_DCHECK(!peer_connection_factory_);
+  RTC_DCHECK(!peer_connection_);
+
+  if (!signaling_thread_.get()) {
+    signaling_thread_ = rtc::Thread::CreateWithSocketServer();
+    signaling_thread_->Start();
+  }
+  peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
+      nullptr /* network_thread */, nullptr /* worker_thread */,
+      signaling_thread_.get(), nullptr /* default_adm */,
+      webrtc::CreateBuiltinAudioEncoderFactory(),
+      webrtc::CreateBuiltinAudioDecoderFactory(),
+      webrtc::CreateBuiltinVideoEncoderFactory(),
+      webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
+      nullptr /* audio_processing */);
+
+  if (!peer_connection_factory_) {
+    main_wnd_->MessageBox("Error", "Failed to initialize PeerConnectionFactory",
+                          true);
+    DeletePeerConnection();
+    return false;
+  }
+
+  if (!CreatePeerConnection()) {
+    main_wnd_->MessageBox("Error", "CreatePeerConnection failed", true);
+    DeletePeerConnection();
+  }
+
+  AddTracks();
+
+  return peer_connection_ != nullptr;
+}
+
+bool Conductor::ReinitializePeerConnectionForLoopback() {
+  loopback_ = true;
+  std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> senders =
+      peer_connection_->GetSenders();
+  peer_connection_ = nullptr;
+  // Loopback is only possible if encryption is disabled.
+  webrtc::PeerConnectionFactoryInterface::Options options;
+  options.disable_encryption = true;
+  peer_connection_factory_->SetOptions(options);
+  if (CreatePeerConnection()) {
+    for (const auto& sender : senders) {
+      peer_connection_->AddTrack(sender->track(), sender->stream_ids());
+    }
+    peer_connection_->CreateOffer(
+        this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
+  }
+  options.disable_encryption = false;
+  peer_connection_factory_->SetOptions(options);
+  return peer_connection_ != nullptr;
+}
+
+bool Conductor::CreatePeerConnection() {
+  RTC_DCHECK(peer_connection_factory_);
+  RTC_DCHECK(!peer_connection_);
+
+  webrtc::PeerConnectionInterface::RTCConfiguration config;
+  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+  webrtc::PeerConnectionInterface::IceServer server;
+  server.uri = GetPeerConnectionString();
+  config.servers.push_back(server);
+
+  webrtc::PeerConnectionDependencies pc_dependencies(this);
+  auto error_or_peer_connection =
+      peer_connection_factory_->CreatePeerConnectionOrError(
+          config, std::move(pc_dependencies));
+  if (error_or_peer_connection.ok()) {
+    peer_connection_ = std::move(error_or_peer_connection.value());
+  }
+  return peer_connection_ != nullptr;
+}
+
+void Conductor::DeletePeerConnection() {
+  main_wnd_->StopLocalRenderer();
+  main_wnd_->StopRemoteRenderer();
+  peer_connection_ = nullptr;
+  peer_connection_factory_ = nullptr;
+  peer_id_ = -1;
+  loopback_ = false;
+}
+
+void Conductor::EnsureStreamingUI() {
+  RTC_DCHECK(peer_connection_);
+  if (main_wnd_->IsWindow()) {
+    if (main_wnd_->current_ui() != MainWindow::STREAMING)
+      main_wnd_->SwitchToStreamingUI();
+  }
+}
+
+//
+// PeerConnectionObserver implementation.
+//
+
+void Conductor::OnAddTrack(
+    rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
+    const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&
+        streams) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id();
+  main_wnd_->QueueUIThreadCallback(NEW_TRACK_ADDED,
+                                   receiver->track().release());
+}
+
+void Conductor::OnRemoveTrack(
+    rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id();
+  main_wnd_->QueueUIThreadCallback(TRACK_REMOVED, receiver->track().release());
+}
+
+void Conductor::OnIceCandidate(const webrtc::IceCandidateInterface* candidate) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << " " << candidate->sdp_mline_index();
+  // For loopback test. To save some connecting delay.
+  if (loopback_) {
+    if (!peer_connection_->AddIceCandidate(candidate)) {
+      RTC_LOG(LS_WARNING) << "Failed to apply the received candidate";
+    }
+    return;
+  }
+
+  Json::Value jmessage;
+  jmessage[kCandidateSdpMidName] = candidate->sdp_mid();
+  jmessage[kCandidateSdpMlineIndexName] = candidate->sdp_mline_index();
+  std::string sdp;
+  if (!candidate->ToString(&sdp)) {
+    RTC_LOG(LS_ERROR) << "Failed to serialize candidate";
+    return;
+  }
+  jmessage[kCandidateSdpName] = sdp;
+
+  Json::StreamWriterBuilder factory;
+  SendMessage(Json::writeString(factory, jmessage));
+}
+
+//
+// PeerConnectionClientObserver implementation.
+//
+
+void Conductor::OnSignedIn() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  main_wnd_->SwitchToPeerList(client_->peers());
+}
+
+void Conductor::OnDisconnected() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+
+  DeletePeerConnection();
+
+  if (main_wnd_->IsWindow())
+    main_wnd_->SwitchToConnectUI();
+}
+
+void Conductor::OnPeerConnected(int id, const std::string& name) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  // Refresh the list if we're showing it.
+  if (main_wnd_->current_ui() == MainWindow::LIST_PEERS)
+    main_wnd_->SwitchToPeerList(client_->peers());
+}
+
+void Conductor::OnPeerDisconnected(int id) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  if (id == peer_id_) {
+    RTC_LOG(LS_INFO) << "Our peer disconnected";
+    main_wnd_->QueueUIThreadCallback(PEER_CONNECTION_CLOSED, NULL);
+  } else {
+    // Refresh the list if we're showing it.
+    if (main_wnd_->current_ui() == MainWindow::LIST_PEERS)
+      main_wnd_->SwitchToPeerList(client_->peers());
+  }
+}
+
+void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) {
+  RTC_DCHECK(peer_id_ == peer_id || peer_id_ == -1);
+  RTC_DCHECK(!message.empty());
+
+  if (!peer_connection_.get()) {
+    RTC_DCHECK(peer_id_ == -1);
+    peer_id_ = peer_id;
+
+    if (!InitializePeerConnection()) {
+      RTC_LOG(LS_ERROR) << "Failed to initialize our PeerConnection instance";
+      client_->SignOut();
+      return;
+    }
+  } else if (peer_id != peer_id_) {
+    RTC_DCHECK(peer_id_ != -1);
+    RTC_LOG(LS_WARNING)
+        << "Received a message from unknown peer while already in a "
+           "conversation with a different peer.";
+    return;
+  }
+
+  Json::CharReaderBuilder factory;
+  std::unique_ptr<Json::CharReader> reader =
+      absl::WrapUnique(factory.newCharReader());
+  Json::Value jmessage;
+  if (!reader->parse(message.data(), message.data() + message.length(),
+                     &jmessage, nullptr)) {
+    RTC_LOG(LS_WARNING) << "Received unknown message. " << message;
+    return;
+  }
+  std::string type_str;
+  std::string json_object;
+
+  rtc::GetStringFromJsonObject(jmessage, kSessionDescriptionTypeName,
+                               &type_str);
+  if (!type_str.empty()) {
+    if (type_str == "offer-loopback") {
+      // This is a loopback call.
+      // Recreate the peerconnection with DTLS disabled.
+      if (!ReinitializePeerConnectionForLoopback()) {
+        RTC_LOG(LS_ERROR) << "Failed to initialize our PeerConnection instance";
+        DeletePeerConnection();
+        client_->SignOut();
+      }
+      return;
+    }
+    absl::optional<webrtc::SdpType> type_maybe =
+        webrtc::SdpTypeFromString(type_str);
+    if (!type_maybe) {
+      RTC_LOG(LS_ERROR) << "Unknown SDP type: " << type_str;
+      return;
+    }
+    webrtc::SdpType type = *type_maybe;
+    std::string sdp;
+    if (!rtc::GetStringFromJsonObject(jmessage, kSessionDescriptionSdpName,
+                                      &sdp)) {
+      RTC_LOG(LS_WARNING)
+          << "Can't parse received session description message.";
+      return;
+    }
+    webrtc::SdpParseError error;
+    std::unique_ptr<webrtc::SessionDescriptionInterface> session_description =
+        webrtc::CreateSessionDescription(type, sdp, &error);
+    if (!session_description) {
+      RTC_LOG(LS_WARNING)
+          << "Can't parse received session description message. "
+             "SdpParseError was: "
+          << error.description;
+      return;
+    }
+    RTC_LOG(LS_INFO) << " Received session description :" << message;
+    peer_connection_->SetRemoteDescription(
+        DummySetSessionDescriptionObserver::Create().get(),
+        session_description.release());
+    if (type == webrtc::SdpType::kOffer) {
+      peer_connection_->CreateAnswer(
+          this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
+    }
+  } else {
+    std::string sdp_mid;
+    int sdp_mlineindex = 0;
+    std::string sdp;
+    if (!rtc::GetStringFromJsonObject(jmessage, kCandidateSdpMidName,
+                                      &sdp_mid) ||
+        !rtc::GetIntFromJsonObject(jmessage, kCandidateSdpMlineIndexName,
+                                   &sdp_mlineindex) ||
+        !rtc::GetStringFromJsonObject(jmessage, kCandidateSdpName, &sdp)) {
+      RTC_LOG(LS_WARNING) << "Can't parse received message.";
+      return;
+    }
+    webrtc::SdpParseError error;
+    std::unique_ptr<webrtc::IceCandidateInterface> candidate(
+        webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, sdp, &error));
+    if (!candidate.get()) {
+      RTC_LOG(LS_WARNING) << "Can't parse received candidate message. "
+                             "SdpParseError was: "
+                          << error.description;
+      return;
+    }
+    if (!peer_connection_->AddIceCandidate(candidate.get())) {
+      RTC_LOG(LS_WARNING) << "Failed to apply the received candidate";
+      return;
+    }
+    RTC_LOG(LS_INFO) << " Received candidate :" << message;
+  }
+}
+
+void Conductor::OnMessageSent(int err) {
+  // Process the next pending message if any.
+  main_wnd_->QueueUIThreadCallback(SEND_MESSAGE_TO_PEER, NULL);
+}
+
+void Conductor::OnServerConnectionFailure() {
+  main_wnd_->MessageBox("Error", ("Failed to connect to " + server_).c_str(),
+                        true);
+}
+
+//
+// MainWndCallback implementation.
+//
+
+void Conductor::StartLogin(const std::string& server, int port) {
+  if (client_->is_connected())
+    return;
+  server_ = server;
+  client_->Connect(server, port, GetPeerName());
+}
+
+void Conductor::DisconnectFromServer() {
+  if (client_->is_connected())
+    client_->SignOut();
+}
+
+void Conductor::ConnectToPeer(int peer_id) {
+  RTC_DCHECK(peer_id_ == -1);
+  RTC_DCHECK(peer_id != -1);
+
+  if (peer_connection_.get()) {
+    main_wnd_->MessageBox(
+        "Error", "We only support connecting to one peer at a time", true);
+    return;
+  }
+
+  if (InitializePeerConnection()) {
+    peer_id_ = peer_id;
+    peer_connection_->CreateOffer(
+        this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
+  } else {
+    main_wnd_->MessageBox("Error", "Failed to initialize PeerConnection", true);
+  }
+}
+
+void Conductor::AddTracks() {
+  if (!peer_connection_->GetSenders().empty()) {
+    return;  // Already added tracks.
+  }
+
+  rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+      peer_connection_factory_->CreateAudioTrack(
+          kAudioLabel,
+          peer_connection_factory_->CreateAudioSource(cricket::AudioOptions())
+              .get()));
+  auto result_or_error = peer_connection_->AddTrack(audio_track, {kStreamId});
+  if (!result_or_error.ok()) {
+    RTC_LOG(LS_ERROR) << "Failed to add audio track to PeerConnection: "
+                      << result_or_error.error().message();
+  }
+
+  rtc::scoped_refptr<CapturerTrackSource> video_device =
+      CapturerTrackSource::Create();
+  if (video_device) {
+    rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track_(
+        peer_connection_factory_->CreateVideoTrack(kVideoLabel,
+                                                   video_device.get()));
+    main_wnd_->StartLocalRenderer(video_track_.get());
+
+    result_or_error = peer_connection_->AddTrack(video_track_, {kStreamId});
+    if (!result_or_error.ok()) {
+      RTC_LOG(LS_ERROR) << "Failed to add video track to PeerConnection: "
+                        << result_or_error.error().message();
+    }
+  } else {
+    RTC_LOG(LS_ERROR) << "OpenVideoCaptureDevice failed";
+  }
+
+  main_wnd_->SwitchToStreamingUI();
+}
+
+void Conductor::DisconnectFromCurrentPeer() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  if (peer_connection_.get()) {
+    client_->SendHangUp(peer_id_);
+    DeletePeerConnection();
+  }
+
+  if (main_wnd_->IsWindow())
+    main_wnd_->SwitchToPeerList(client_->peers());
+}
+
+void Conductor::UIThreadCallback(int msg_id, void* data) {
+  switch (msg_id) {
+    case PEER_CONNECTION_CLOSED:
+      RTC_LOG(LS_INFO) << "PEER_CONNECTION_CLOSED";
+      DeletePeerConnection();
+
+      if (main_wnd_->IsWindow()) {
+        if (client_->is_connected()) {
+          main_wnd_->SwitchToPeerList(client_->peers());
+        } else {
+          main_wnd_->SwitchToConnectUI();
+        }
+      } else {
+        DisconnectFromServer();
+      }
+      break;
+
+    case SEND_MESSAGE_TO_PEER: {
+      RTC_LOG(LS_INFO) << "SEND_MESSAGE_TO_PEER";
+      std::string* msg = reinterpret_cast<std::string*>(data);
+      if (msg) {
+        // For convenience, we always run the message through the queue.
+        // This way we can be sure that messages are sent to the server
+        // in the same order they were signaled without much hassle.
+        pending_messages_.push_back(msg);
+      }
+
+      if (!pending_messages_.empty() && !client_->IsSendingMessage()) {
+        msg = pending_messages_.front();
+        pending_messages_.pop_front();
+
+        if (!client_->SendToPeer(peer_id_, *msg) && peer_id_ != -1) {
+          RTC_LOG(LS_ERROR) << "SendToPeer failed";
+          DisconnectFromServer();
+        }
+        delete msg;
+      }
+
+      if (!peer_connection_.get())
+        peer_id_ = -1;
+
+      break;
+    }
+
+    case NEW_TRACK_ADDED: {
+      auto* track = reinterpret_cast<webrtc::MediaStreamTrackInterface*>(data);
+      if (track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) {
+        auto* video_track = static_cast<webrtc::VideoTrackInterface*>(track);
+        main_wnd_->StartRemoteRenderer(video_track);
+      }
+      track->Release();
+      break;
+    }
+
+    case TRACK_REMOVED: {
+      // Remote peer stopped sending a track.
+      auto* track = reinterpret_cast<webrtc::MediaStreamTrackInterface*>(data);
+      track->Release();
+      break;
+    }
+
+    default:
+      RTC_DCHECK_NOTREACHED();
+      break;
+  }
+}
+
+void Conductor::OnSuccess(webrtc::SessionDescriptionInterface* desc) {
+  peer_connection_->SetLocalDescription(
+      DummySetSessionDescriptionObserver::Create().get(), desc);
+
+  std::string sdp;
+  desc->ToString(&sdp);
+
+  // For loopback test. To save some connecting delay.
+  if (loopback_) {
+    // Replace message type from "offer" to "answer"
+    std::unique_ptr<webrtc::SessionDescriptionInterface> session_description =
+        webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp);
+    peer_connection_->SetRemoteDescription(
+        DummySetSessionDescriptionObserver::Create().get(),
+        session_description.release());
+    return;
+  }
+
+  Json::Value jmessage;
+  jmessage[kSessionDescriptionTypeName] =
+      webrtc::SdpTypeToString(desc->GetType());
+  jmessage[kSessionDescriptionSdpName] = sdp;
+
+  Json::StreamWriterBuilder factory;
+  SendMessage(Json::writeString(factory, jmessage));
+}
+
+void Conductor::OnFailure(webrtc::RTCError error) {
+  RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message();
+}
+
+void Conductor::SendMessage(const std::string& json_object) {
+  std::string* msg = new std::string(json_object);
+  main_wnd_->QueueUIThreadCallback(SEND_MESSAGE_TO_PEER, msg);
+}
diff --git a/third_party/libwebrtc/examples/peerconnection/client/conductor.h b/third_party/libwebrtc/examples/peerconnection/client/conductor.h
new file mode 100644
index 0000000000..80617d3cf4
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/client/conductor.h
@@ -0,0 +1,136 @@
+/*
+ *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_PEERCONNECTION_CLIENT_CONDUCTOR_H_
+#define EXAMPLES_PEERCONNECTION_CLIENT_CONDUCTOR_H_
+
+#include <deque>
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/media_stream_interface.h"
+#include "api/peer_connection_interface.h"
+#include "examples/peerconnection/client/main_wnd.h"
+#include "examples/peerconnection/client/peer_connection_client.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+class VideoCaptureModule;
+}  // namespace webrtc
+
+namespace cricket {
+class VideoRenderer;
+}  // namespace cricket
+
+class Conductor : public webrtc::PeerConnectionObserver,
+                  public webrtc::CreateSessionDescriptionObserver,
+                  public PeerConnectionClientObserver,
+                  public MainWndCallback {
+ public:
+  enum CallbackID {
+    MEDIA_CHANNELS_INITIALIZED = 1,
+    PEER_CONNECTION_CLOSED,
+    SEND_MESSAGE_TO_PEER,
+    NEW_TRACK_ADDED,
+    TRACK_REMOVED,
+  };
+
+  Conductor(PeerConnectionClient* client, MainWindow* main_wnd);
+
+  bool connection_active() const;
+
+  void Close() override;
+
+ protected:
+  ~Conductor();
+  bool InitializePeerConnection();
+  bool ReinitializePeerConnectionForLoopback();
+  bool CreatePeerConnection();
+  void DeletePeerConnection();
+  void EnsureStreamingUI();
+  void AddTracks();
+
+  //
+  // PeerConnectionObserver implementation.
+  //
+
+  void OnSignalingChange(
+      webrtc::PeerConnectionInterface::SignalingState new_state) override {}
+  void OnAddTrack(
+      rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
+      const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&
+          streams) override;
+  void OnRemoveTrack(
+      rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) override;
+  void OnDataChannel(
+      rtc::scoped_refptr<webrtc::DataChannelInterface> channel) override {}
+  void OnRenegotiationNeeded() override {}
+  void OnIceConnectionChange(
+      webrtc::PeerConnectionInterface::IceConnectionState new_state) override {}
+  void OnIceGatheringChange(
+      webrtc::PeerConnectionInterface::IceGatheringState new_state) override {}
+  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
+  void OnIceConnectionReceivingChange(bool receiving) override {}
+
+  //
+  // PeerConnectionClientObserver implementation.
+  //
+
+  void OnSignedIn() override;
+
+  void OnDisconnected() override;
+
+  void OnPeerConnected(int id, const std::string& name) override;
+
+  void OnPeerDisconnected(int id) override;
+
+  void OnMessageFromPeer(int peer_id, const std::string& message) override;
+
+  void OnMessageSent(int err) override;
+
+  void OnServerConnectionFailure() override;
+
+  //
+  // MainWndCallback implementation.
+  //
+
+  void StartLogin(const std::string& server, int port) override;
+
+  void DisconnectFromServer() override;
+
+  void ConnectToPeer(int peer_id) override;
+
+  void DisconnectFromCurrentPeer() override;
+
+  void UIThreadCallback(int msg_id, void* data) override;
+
+  // CreateSessionDescriptionObserver implementation.
+  void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
+  void OnFailure(webrtc::RTCError error) override;
+
+ protected:
+  // Send a message to the remote peer.
+  void SendMessage(const std::string& json_object);
+
+  int peer_id_;
+  bool loopback_;
+  std::unique_ptr<rtc::Thread> signaling_thread_;
+  rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
+  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
+      peer_connection_factory_;
+  PeerConnectionClient* client_;
+  MainWindow* main_wnd_;
+  std::deque<std::string*> pending_messages_;
+  std::string server_;
+};
+
+#endif  // EXAMPLES_PEERCONNECTION_CLIENT_CONDUCTOR_H_
diff --git a/third_party/libwebrtc/examples/peerconnection/client/defaults.cc b/third_party/libwebrtc/examples/peerconnection/client/defaults.cc
new file mode 100644
index 0000000000..ee3a9e1f0a
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/client/defaults.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/peerconnection/client/defaults.h"
+
+#include <stdlib.h>
+
+#ifdef WIN32
+#include <winsock2.h>
+#else
+#include <unistd.h>
+#endif
+
+#include "rtc_base/arraysize.h"
+
+const char kAudioLabel[] = "audio_label";
+const char kVideoLabel[] = "video_label";
+const char kStreamId[] = "stream_id";
+const uint16_t kDefaultServerPort = 8888;
+
+std::string GetEnvVarOrDefault(const char* env_var_name,
+                               const char* default_value) {
+  std::string value;
+  const char* env_var = getenv(env_var_name);
+  if (env_var)
+    value = env_var;
+
+  if (value.empty())
+    value = default_value;
+
+  return value;
+}
+
+std::string GetPeerConnectionString() {
+  return GetEnvVarOrDefault("WEBRTC_CONNECT", "stun:stun.l.google.com:19302");
+}
+
+std::string GetDefaultServerName() {
+  return GetEnvVarOrDefault("WEBRTC_SERVER", "localhost");
+}
+
+std::string GetPeerName() {
+  char computer_name[256];
+  std::string ret(GetEnvVarOrDefault("USERNAME", "user"));
+  ret += '@';
+  if (gethostname(computer_name, arraysize(computer_name)) == 0) {
+    ret += computer_name;
+  } else {
+    ret += "host";
+  }
+  return ret;
+}
diff --git a/third_party/libwebrtc/examples/peerconnection/client/defaults.h b/third_party/libwebrtc/examples/peerconnection/client/defaults.h
new file mode 100644
index 0000000000..30936fd9d4
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/client/defaults.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_PEERCONNECTION_CLIENT_DEFAULTS_H_
+#define EXAMPLES_PEERCONNECTION_CLIENT_DEFAULTS_H_
+
+#include <stdint.h>
+
+#include <string>
+
+extern const char kAudioLabel[];
+extern const char kVideoLabel[];
+extern const char kStreamId[];
+extern const uint16_t kDefaultServerPort;
+
+std::string GetEnvVarOrDefault(const char* env_var_name,
+                               const char* default_value);
+std::string GetPeerConnectionString();
+std::string GetDefaultServerName();
+std::string GetPeerName();
+
+#endif  // EXAMPLES_PEERCONNECTION_CLIENT_DEFAULTS_H_
diff --git a/third_party/libwebrtc/examples/peerconnection/client/flag_defs.h b/third_party/libwebrtc/examples/peerconnection/client/flag_defs.h
new file mode 100644
index 0000000000..986daf64ce
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/client/flag_defs.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_PEERCONNECTION_CLIENT_FLAG_DEFS_H_
+#define EXAMPLES_PEERCONNECTION_CLIENT_FLAG_DEFS_H_
+
+#include <string>
+
+#include "absl/flags/flag.h"
+
+extern const uint16_t kDefaultServerPort;  // From defaults.[h|cc]
+
+// Define flags for the peerconnect_client testing tool, in a separate
+// header file so that they can be shared across the different main.cc's
+// for each platform.
+
+ABSL_FLAG(bool,
+          autoconnect,
+          false,
+          "Connect to the server without user "
+          "intervention.");
+ABSL_FLAG(std::string, server, "localhost", "The server to connect to.");
+ABSL_FLAG(int,
+          port,
+          kDefaultServerPort,
+          "The port on which the server is listening.");
+ABSL_FLAG(
+    bool,
+    autocall,
+    false,
+    "Call the first available other client on "
+    "the server without user intervention.  Note: this flag should only be set "
+    "to true on one of the two clients.");
+
+ABSL_FLAG(
+    std::string,
+    force_fieldtrials,
+    "",
+    "Field trials control experimental features. This flag specifies the field "
+    "trials in effect. E.g. running with "
+    "--force_fieldtrials=WebRTC-FooFeature/Enabled/ "
+    "will assign the group Enabled to field trial WebRTC-FooFeature. Multiple "
+    "trials are separated by \"/\"");
+
+#endif  // EXAMPLES_PEERCONNECTION_CLIENT_FLAG_DEFS_H_
diff --git a/third_party/libwebrtc/examples/peerconnection/client/linux/main.cc b/third_party/libwebrtc/examples/peerconnection/client/linux/main.cc
new file mode 100644
index 0000000000..ad3d671073
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/client/linux/main.cc
@@ -0,0 +1,121 @@
+/*
+ *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <glib.h>
+#include <gtk/gtk.h>
+#include <stdio.h>
+
+#include "absl/flags/parse.h"
+#include "api/scoped_refptr.h"
+#include "examples/peerconnection/client/conductor.h"
+#include "examples/peerconnection/client/flag_defs.h"
+#include "examples/peerconnection/client/linux/main_wnd.h"
+#include "examples/peerconnection/client/peer_connection_client.h"
+#include "rtc_base/physical_socket_server.h"
+#include "rtc_base/ssl_adapter.h"
+#include "rtc_base/thread.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/field_trial.h"
+
+class CustomSocketServer : public rtc::PhysicalSocketServer {
+ public:
+  explicit CustomSocketServer(GtkMainWnd* wnd)
+      : wnd_(wnd), conductor_(NULL), client_(NULL) {}
+  virtual ~CustomSocketServer() {}
+
+  void SetMessageQueue(rtc::Thread* queue) override { message_queue_ = queue; }
+
+  void set_client(PeerConnectionClient* client) { client_ = client; }
+  void set_conductor(Conductor* conductor) { conductor_ = conductor; }
+
+  // Override so that we can also pump the GTK message loop.
+  // This function never waits.
+  bool Wait(webrtc::TimeDelta max_wait_duration, bool process_io) override {
+    // Pump GTK events.
+    // TODO(henrike): We really should move either the socket server or UI to a
+    // different thread.  Alternatively we could look at merging the two loops
+    // by implementing a dispatcher for the socket server and/or use
+    // g_main_context_set_poll_func.
+    while (gtk_events_pending())
+      gtk_main_iteration();
+
+    if (!wnd_->IsWindow() && !conductor_->connection_active() &&
+        client_ != NULL && !client_->is_connected()) {
+      message_queue_->Quit();
+    }
+    return rtc::PhysicalSocketServer::Wait(webrtc::TimeDelta::Zero(),
+                                           process_io);
+  }
+
+ protected:
+  rtc::Thread* message_queue_;
+  GtkMainWnd* wnd_;
+  Conductor* conductor_;
+  PeerConnectionClient* client_;
+};
+
+int main(int argc, char* argv[]) {
+  gtk_init(&argc, &argv);
+// g_type_init API is deprecated (and does nothing) since glib 2.35.0, see:
+// https://mail.gnome.org/archives/commits-list/2012-November/msg07809.html
+#if !GLIB_CHECK_VERSION(2, 35, 0)
+  g_type_init();
+#endif
+// g_thread_init API is deprecated since glib 2.31.0, see release note:
+// http://mail.gnome.org/archives/gnome-announce-list/2011-October/msg00041.html
+#if !GLIB_CHECK_VERSION(2, 31, 0)
+  g_thread_init(NULL);
+#endif
+
+  absl::ParseCommandLine(argc, argv);
+
+  // InitFieldTrialsFromString stores the char*, so the char array must outlive
+  // the application.
+  const std::string forced_field_trials =
+      absl::GetFlag(FLAGS_force_fieldtrials);
+  webrtc::field_trial::InitFieldTrialsFromString(forced_field_trials.c_str());
+
+  // Abort if the user specifies a port that is outside the allowed
+  // range [1, 65535].
+  if ((absl::GetFlag(FLAGS_port) < 1) || (absl::GetFlag(FLAGS_port) > 65535)) {
+    printf("Error: %i is not a valid port.\n", absl::GetFlag(FLAGS_port));
+    return -1;
+  }
+
+  const std::string server = absl::GetFlag(FLAGS_server);
+  GtkMainWnd wnd(server.c_str(), absl::GetFlag(FLAGS_port),
+                 absl::GetFlag(FLAGS_autoconnect),
+                 absl::GetFlag(FLAGS_autocall));
+  wnd.Create();
+
+  CustomSocketServer socket_server(&wnd);
+  rtc::AutoSocketServerThread thread(&socket_server);
+
+  rtc::InitializeSSL();
+  // Must be constructed after we set the socketserver.
+  PeerConnectionClient client;
+  auto conductor = rtc::make_ref_counted<Conductor>(&client, &wnd);
+  socket_server.set_client(&client);
+  socket_server.set_conductor(conductor.get());
+
+  thread.Run();
+
+  // gtk_main();
+  wnd.Destroy();
+
+  // TODO(henrike): Run the Gtk main loop to tear down the connection.
+  /*
+  while (gtk_events_pending()) {
+    gtk_main_iteration();
+  }
+  */
+  rtc::CleanupSSL();
+  return 0;
+}
diff --git a/third_party/libwebrtc/examples/peerconnection/client/linux/main_wnd.cc b/third_party/libwebrtc/examples/peerconnection/client/linux/main_wnd.cc
new file mode 100644
index 0000000000..2be75d8f8d
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/client/linux/main_wnd.cc
@@ -0,0 +1,545 @@
+/*
+ *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/peerconnection/client/linux/main_wnd.h"
+
+#include <cairo.h>
+#include <gdk/gdk.h>
+#include <gdk/gdkkeysyms.h>
+#include <glib-object.h>
+#include <glib.h>
+#include <gobject/gclosure.h>
+#include <gtk/gtk.h>
+#include <stddef.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <cstdint>
+#include <map>
+#include <utility>
+
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video/video_rotation.h"
+#include "api/video/video_source_interface.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "third_party/libyuv/include/libyuv/convert.h"
+#include "third_party/libyuv/include/libyuv/convert_from.h"
+
+namespace {
+
+//
+// Simple static functions that simply forward the callback to the
+// GtkMainWnd instance.
+//
+
+gboolean OnDestroyedCallback(GtkWidget* widget,
+                             GdkEvent* event,
+                             gpointer data) {
+  reinterpret_cast<GtkMainWnd*>(data)->OnDestroyed(widget, event);
+  return FALSE;
+}
+
+void OnClickedCallback(GtkWidget* widget, gpointer data) {
+  reinterpret_cast<GtkMainWnd*>(data)->OnClicked(widget);
+}
+
+gboolean SimulateButtonClick(gpointer button) {
+  g_signal_emit_by_name(button, "clicked");
+  return false;
+}
+
+gboolean OnKeyPressCallback(GtkWidget* widget,
+                            GdkEventKey* key,
+                            gpointer data) {
+  reinterpret_cast<GtkMainWnd*>(data)->OnKeyPress(widget, key);
+  return false;
+}
+
+void OnRowActivatedCallback(GtkTreeView* tree_view,
+                            GtkTreePath* path,
+                            GtkTreeViewColumn* column,
+                            gpointer data) {
+  reinterpret_cast<GtkMainWnd*>(data)->OnRowActivated(tree_view, path, column);
+}
+
+gboolean SimulateLastRowActivated(gpointer data) {
+  GtkTreeView* tree_view = reinterpret_cast<GtkTreeView*>(data);
+  GtkTreeModel* model = gtk_tree_view_get_model(tree_view);
+
+  // "if iter is NULL, then the number of toplevel nodes is returned."
+  int rows = gtk_tree_model_iter_n_children(model, NULL);
+  GtkTreePath* lastpath = gtk_tree_path_new_from_indices(rows - 1, -1);
+
+  // Select the last item in the list
+  GtkTreeSelection* selection = gtk_tree_view_get_selection(tree_view);
+  gtk_tree_selection_select_path(selection, lastpath);
+
+  // Our TreeView only has one column, so it is column 0.
+  GtkTreeViewColumn* column = gtk_tree_view_get_column(tree_view, 0);
+
+  gtk_tree_view_row_activated(tree_view, lastpath, column);
+
+  gtk_tree_path_free(lastpath);
+  return false;
+}
+
+// Creates a tree view, that we use to display the list of peers.
+void InitializeList(GtkWidget* list) {
+  GtkCellRenderer* renderer = gtk_cell_renderer_text_new();
+  GtkTreeViewColumn* column = gtk_tree_view_column_new_with_attributes(
+      "List Items", renderer, "text", 0, NULL);
+  gtk_tree_view_append_column(GTK_TREE_VIEW(list), column);
+  GtkListStore* store = gtk_list_store_new(2, G_TYPE_STRING, G_TYPE_INT);
+  gtk_tree_view_set_model(GTK_TREE_VIEW(list), GTK_TREE_MODEL(store));
+  g_object_unref(store);
+}
+
+// Adds an entry to a tree view.
+void AddToList(GtkWidget* list, const gchar* str, int value) {
+  GtkListStore* store =
+      GTK_LIST_STORE(gtk_tree_view_get_model(GTK_TREE_VIEW(list)));
+
+  GtkTreeIter iter;
+  gtk_list_store_append(store, &iter);
+  gtk_list_store_set(store, &iter, 0, str, 1, value, -1);
+}
+
+struct UIThreadCallbackData {
+  explicit UIThreadCallbackData(MainWndCallback* cb, int id, void* d)
+      : callback(cb), msg_id(id), data(d) {}
+  MainWndCallback* callback;
+  int msg_id;
+  void* data;
+};
+
+gboolean HandleUIThreadCallback(gpointer data) {
+  UIThreadCallbackData* cb_data = reinterpret_cast<UIThreadCallbackData*>(data);
+  cb_data->callback->UIThreadCallback(cb_data->msg_id, cb_data->data);
+  delete cb_data;
+  return false;
+}
+
+gboolean Redraw(gpointer data) {
+  GtkMainWnd* wnd = reinterpret_cast<GtkMainWnd*>(data);
+  wnd->OnRedraw();
+  return false;
+}
+
+gboolean Draw(GtkWidget* widget, cairo_t* cr, gpointer data) {
+  GtkMainWnd* wnd = reinterpret_cast<GtkMainWnd*>(data);
+  wnd->Draw(widget, cr);
+  return false;
+}
+
+}  // namespace
+
+//
+// GtkMainWnd implementation.
+//
+
+GtkMainWnd::GtkMainWnd(const char* server,
+                       int port,
+                       bool autoconnect,
+                       bool autocall)
+    : window_(NULL),
+      draw_area_(NULL),
+      vbox_(NULL),
+      server_edit_(NULL),
+      port_edit_(NULL),
+      peer_list_(NULL),
+      callback_(NULL),
+      server_(server),
+      autoconnect_(autoconnect),
+      autocall_(autocall) {
+  char buffer[10];
+  snprintf(buffer, sizeof(buffer), "%i", port);
+  port_ = buffer;
+}
+
+GtkMainWnd::~GtkMainWnd() {
+  RTC_DCHECK(!IsWindow());
+}
+
+void GtkMainWnd::RegisterObserver(MainWndCallback* callback) {
+  callback_ = callback;
+}
+
+bool GtkMainWnd::IsWindow() {
+  return window_ != NULL && GTK_IS_WINDOW(window_);
+}
+
+void GtkMainWnd::MessageBox(const char* caption,
+                            const char* text,
+                            bool is_error) {
+  GtkWidget* dialog = gtk_message_dialog_new(
+      GTK_WINDOW(window_), GTK_DIALOG_DESTROY_WITH_PARENT,
+      is_error ? GTK_MESSAGE_ERROR : GTK_MESSAGE_INFO, GTK_BUTTONS_CLOSE, "%s",
+      text);
+  gtk_window_set_title(GTK_WINDOW(dialog), caption);
+  gtk_dialog_run(GTK_DIALOG(dialog));
+  gtk_widget_destroy(dialog);
+}
+
+MainWindow::UI GtkMainWnd::current_ui() {
+  if (vbox_)
+    return CONNECT_TO_SERVER;
+
+  if (peer_list_)
+    return LIST_PEERS;
+
+  return STREAMING;
+}
+
+void GtkMainWnd::StartLocalRenderer(webrtc::VideoTrackInterface* local_video) {
+  local_renderer_.reset(new VideoRenderer(this, local_video));
+}
+
+void GtkMainWnd::StopLocalRenderer() {
+  local_renderer_.reset();
+}
+
+void GtkMainWnd::StartRemoteRenderer(
+    webrtc::VideoTrackInterface* remote_video) {
+  remote_renderer_.reset(new VideoRenderer(this, remote_video));
+}
+
+void GtkMainWnd::StopRemoteRenderer() {
+  remote_renderer_.reset();
+}
+
+void GtkMainWnd::QueueUIThreadCallback(int msg_id, void* data) {
+  g_idle_add(HandleUIThreadCallback,
+             new UIThreadCallbackData(callback_, msg_id, data));
+}
+
+bool GtkMainWnd::Create() {
+  RTC_DCHECK(window_ == NULL);
+
+  window_ = gtk_window_new(GTK_WINDOW_TOPLEVEL);
+  if (window_) {
+    gtk_window_set_position(GTK_WINDOW(window_), GTK_WIN_POS_CENTER);
+    gtk_window_set_default_size(GTK_WINDOW(window_), 640, 480);
+    gtk_window_set_title(GTK_WINDOW(window_), "PeerConnection client");
+    g_signal_connect(G_OBJECT(window_), "delete-event",
+                     G_CALLBACK(&OnDestroyedCallback), this);
+    g_signal_connect(window_, "key-press-event", G_CALLBACK(OnKeyPressCallback),
+                     this);
+
+    SwitchToConnectUI();
+  }
+
+  return window_ != NULL;
+}
+
+bool GtkMainWnd::Destroy() {
+  if (!IsWindow())
+    return false;
+
+  gtk_widget_destroy(window_);
+  window_ = NULL;
+
+  return true;
+}
+
+void GtkMainWnd::SwitchToConnectUI() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+
+  RTC_DCHECK(IsWindow());
+  RTC_DCHECK(vbox_ == NULL);
+
+  gtk_container_set_border_width(GTK_CONTAINER(window_), 10);
+
+  if (peer_list_) {
+    gtk_widget_destroy(peer_list_);
+    peer_list_ = NULL;
+  }
+
+  vbox_ = gtk_box_new(GTK_ORIENTATION_VERTICAL, 5);
+  GtkWidget* valign = gtk_alignment_new(0, 1, 0, 0);
+  gtk_container_add(GTK_CONTAINER(vbox_), valign);
+  gtk_container_add(GTK_CONTAINER(window_), vbox_);
+
+  GtkWidget* hbox = gtk_box_new(GTK_ORIENTATION_HORIZONTAL, 5);
+
+  GtkWidget* label = gtk_label_new("Server");
+  gtk_container_add(GTK_CONTAINER(hbox), label);
+
+  server_edit_ = gtk_entry_new();
+  gtk_entry_set_text(GTK_ENTRY(server_edit_), server_.c_str());
+  gtk_widget_set_size_request(server_edit_, 400, 30);
+  gtk_container_add(GTK_CONTAINER(hbox), server_edit_);
+
+  port_edit_ = gtk_entry_new();
+  gtk_entry_set_text(GTK_ENTRY(port_edit_), port_.c_str());
+  gtk_widget_set_size_request(port_edit_, 70, 30);
+  gtk_container_add(GTK_CONTAINER(hbox), port_edit_);
+
+  GtkWidget* button = gtk_button_new_with_label("Connect");
+  gtk_widget_set_size_request(button, 70, 30);
+  g_signal_connect(button, "clicked", G_CALLBACK(OnClickedCallback), this);
+  gtk_container_add(GTK_CONTAINER(hbox), button);
+
+  GtkWidget* halign = gtk_alignment_new(1, 0, 0, 0);
+  gtk_container_add(GTK_CONTAINER(halign), hbox);
+  gtk_box_pack_start(GTK_BOX(vbox_), halign, FALSE, FALSE, 0);
+
+  gtk_widget_show_all(window_);
+
+  if (autoconnect_)
+    g_idle_add(SimulateButtonClick, button);
+}
+
+void GtkMainWnd::SwitchToPeerList(const Peers& peers) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+
+  if (!peer_list_) {
+    gtk_container_set_border_width(GTK_CONTAINER(window_), 0);
+    if (vbox_) {
+      gtk_widget_destroy(vbox_);
+      vbox_ = NULL;
+      server_edit_ = NULL;
+      port_edit_ = NULL;
+    } else if (draw_area_) {
+      gtk_widget_destroy(draw_area_);
+      draw_area_ = NULL;
+      draw_buffer_.reset();
+    }
+
+    peer_list_ = gtk_tree_view_new();
+    g_signal_connect(peer_list_, "row-activated",
+                     G_CALLBACK(OnRowActivatedCallback), this);
+    gtk_tree_view_set_headers_visible(GTK_TREE_VIEW(peer_list_), FALSE);
+    InitializeList(peer_list_);
+    gtk_container_add(GTK_CONTAINER(window_), peer_list_);
+    gtk_widget_show_all(window_);
+  } else {
+    GtkListStore* store =
+        GTK_LIST_STORE(gtk_tree_view_get_model(GTK_TREE_VIEW(peer_list_)));
+    gtk_list_store_clear(store);
+  }
+
+  AddToList(peer_list_, "List of currently connected peers:", -1);
+  for (Peers::const_iterator i = peers.begin(); i != peers.end(); ++i)
+    AddToList(peer_list_, i->second.c_str(), i->first);
+
+  if (autocall_ && peers.begin() != peers.end())
+    g_idle_add(SimulateLastRowActivated, peer_list_);
+}
+
+void GtkMainWnd::SwitchToStreamingUI() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+
+  RTC_DCHECK(draw_area_ == NULL);
+
+  gtk_container_set_border_width(GTK_CONTAINER(window_), 0);
+  if (peer_list_) {
+    gtk_widget_destroy(peer_list_);
+    peer_list_ = NULL;
+  }
+
+  draw_area_ = gtk_drawing_area_new();
+  gtk_container_add(GTK_CONTAINER(window_), draw_area_);
+  g_signal_connect(G_OBJECT(draw_area_), "draw", G_CALLBACK(&::Draw), this);
+
+  gtk_widget_show_all(window_);
+}
+
+void GtkMainWnd::OnDestroyed(GtkWidget* widget, GdkEvent* event) {
+  callback_->Close();
+  window_ = NULL;
+  draw_area_ = NULL;
+  vbox_ = NULL;
+  server_edit_ = NULL;
+  port_edit_ = NULL;
+  peer_list_ = NULL;
+}
+
+void GtkMainWnd::OnClicked(GtkWidget* widget) {
+  // Make the connect button insensitive, so that it cannot be clicked more than
+  // once.  Now that the connection includes auto-retry, it should not be
+  // necessary to click it more than once.
+  gtk_widget_set_sensitive(widget, false);
+  server_ = gtk_entry_get_text(GTK_ENTRY(server_edit_));
+  port_ = gtk_entry_get_text(GTK_ENTRY(port_edit_));
+  int port = port_.length() ? atoi(port_.c_str()) : 0;
+  callback_->StartLogin(server_, port);
+}
+
+void GtkMainWnd::OnKeyPress(GtkWidget* widget, GdkEventKey* key) {
+  if (key->type == GDK_KEY_PRESS) {
+    switch (key->keyval) {
+      case GDK_KEY_Escape:
+        if (draw_area_) {
+          callback_->DisconnectFromCurrentPeer();
+        } else if (peer_list_) {
+          callback_->DisconnectFromServer();
+        }
+        break;
+
+      case GDK_KEY_KP_Enter:
+      case GDK_KEY_Return:
+        if (vbox_) {
+          OnClicked(NULL);
+        } else if (peer_list_) {
+          // OnRowActivated will be called automatically when the user
+          // presses enter.
+        }
+        break;
+
+      default:
+        break;
+    }
+  }
+}
+
+void GtkMainWnd::OnRowActivated(GtkTreeView* tree_view,
+                                GtkTreePath* path,
+                                GtkTreeViewColumn* column) {
+  RTC_DCHECK(peer_list_ != NULL);
+  GtkTreeIter iter;
+  GtkTreeModel* model;
+  GtkTreeSelection* selection =
+      gtk_tree_view_get_selection(GTK_TREE_VIEW(tree_view));
+  if (gtk_tree_selection_get_selected(selection, &model, &iter)) {
+    char* text;
+    int id = -1;
+    gtk_tree_model_get(model, &iter, 0, &text, 1, &id, -1);
+    if (id != -1)
+      callback_->ConnectToPeer(id);
+    g_free(text);
+  }
+}
+
+void GtkMainWnd::OnRedraw() {
+  gdk_threads_enter();
+
+  VideoRenderer* remote_renderer = remote_renderer_.get();
+  if (remote_renderer && remote_renderer->image() != NULL &&
+      draw_area_ != NULL) {
+    width_ = remote_renderer->width();
+    height_ = remote_renderer->height();
+
+    if (!draw_buffer_.get()) {
+      draw_buffer_size_ = (width_ * height_ * 4) * 4;
+      draw_buffer_.reset(new uint8_t[draw_buffer_size_]);
+      gtk_widget_set_size_request(draw_area_, width_ * 2, height_ * 2);
+    }
+
+    const uint32_t* image =
+        reinterpret_cast<const uint32_t*>(remote_renderer->image());
+    uint32_t* scaled = reinterpret_cast<uint32_t*>(draw_buffer_.get());
+    for (int r = 0; r < height_; ++r) {
+      for (int c = 0; c < width_; ++c) {
+        int x = c * 2;
+        scaled[x] = scaled[x + 1] = image[c];
+      }
+
+      uint32_t* prev_line = scaled;
+      scaled += width_ * 2;
+      memcpy(scaled, prev_line, (width_ * 2) * 4);
+
+      image += width_;
+      scaled += width_ * 2;
+    }
+
+    VideoRenderer* local_renderer = local_renderer_.get();
+    if (local_renderer && local_renderer->image()) {
+      image = reinterpret_cast<const uint32_t*>(local_renderer->image());
+      scaled = reinterpret_cast<uint32_t*>(draw_buffer_.get());
+      // Position the local preview on the right side.
+      scaled += (width_ * 2) - (local_renderer->width() / 2);
+      // right margin...
+      scaled -= 10;
+      // ... towards the bottom.
+      scaled += (height_ * width_ * 4) - ((local_renderer->height() / 2) *
+                                          (local_renderer->width() / 2) * 4);
+      // bottom margin...
+      scaled -= (width_ * 2) * 5;
+      for (int r = 0; r < local_renderer->height(); r += 2) {
+        for (int c = 0; c < local_renderer->width(); c += 2) {
+          scaled[c / 2] = image[c + r * local_renderer->width()];
+        }
+        scaled += width_ * 2;
+      }
+    }
+
+    gtk_widget_queue_draw(draw_area_);
+  }
+
+  gdk_threads_leave();
+}
+
+void GtkMainWnd::Draw(GtkWidget* widget, cairo_t* cr) {
+  cairo_format_t format = CAIRO_FORMAT_ARGB32;
+  cairo_surface_t* surface = cairo_image_surface_create_for_data(
+      draw_buffer_.get(), format, width_ * 2, height_ * 2,
+      cairo_format_stride_for_width(format, width_ * 2));
+  cairo_set_source_surface(cr, surface, 0, 0);
+  cairo_rectangle(cr, 0, 0, width_ * 2, height_ * 2);
+  cairo_fill(cr);
+  cairo_surface_destroy(surface);
+}
+
+GtkMainWnd::VideoRenderer::VideoRenderer(
+    GtkMainWnd* main_wnd,
+    webrtc::VideoTrackInterface* track_to_render)
+    : width_(0),
+      height_(0),
+      main_wnd_(main_wnd),
+      rendered_track_(track_to_render) {
+  rendered_track_->AddOrUpdateSink(this, rtc::VideoSinkWants());
+}
+
+GtkMainWnd::VideoRenderer::~VideoRenderer() {
+  rendered_track_->RemoveSink(this);
+}
+
+void GtkMainWnd::VideoRenderer::SetSize(int width, int height) {
+  gdk_threads_enter();
+
+  if (width_ == width && height_ == height) {
+    return;
+  }
+
+  width_ = width;
+  height_ = height;
+  image_.reset(new uint8_t[width * height * 4]);
+  gdk_threads_leave();
+}
+
+void GtkMainWnd::VideoRenderer::OnFrame(const webrtc::VideoFrame& video_frame) {
+  gdk_threads_enter();
+
+  rtc::scoped_refptr<webrtc::I420BufferInterface> buffer(
+      video_frame.video_frame_buffer()->ToI420());
+  if (video_frame.rotation() != webrtc::kVideoRotation_0) {
+    buffer = webrtc::I420Buffer::Rotate(*buffer, video_frame.rotation());
+  }
+  SetSize(buffer->width(), buffer->height());
+
+  // TODO(bugs.webrtc.org/6857): This conversion is correct for little-endian
+  // only. Cairo ARGB32 treats pixels as 32-bit values in *native* byte order,
+  // with B in the least significant byte of the 32-bit value. Which on
+  // little-endian means that memory layout is BGRA, with the B byte stored at
+  // lowest address. Libyuv's ARGB format (surprisingly?) uses the same
+  // little-endian format, with B in the first byte in memory, regardless of
+  // native endianness.
+  libyuv::I420ToARGB(buffer->DataY(), buffer->StrideY(), buffer->DataU(),
+                     buffer->StrideU(), buffer->DataV(), buffer->StrideV(),
+                     image_.get(), width_ * 4, buffer->width(),
+                     buffer->height());
+
+  gdk_threads_leave();
+
+  g_idle_add(Redraw, main_wnd_);
+}
diff --git a/third_party/libwebrtc/examples/peerconnection/client/linux/main_wnd.h b/third_party/libwebrtc/examples/peerconnection/client/linux/main_wnd.h
new file mode 100644
index 0000000000..3b31e1be3b
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/client/linux/main_wnd.h
@@ -0,0 +1,128 @@
+/*
+ *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_PEERCONNECTION_CLIENT_LINUX_MAIN_WND_H_
+#define EXAMPLES_PEERCONNECTION_CLIENT_LINUX_MAIN_WND_H_
+
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+
+#include "api/media_stream_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "examples/peerconnection/client/main_wnd.h"
+#include "examples/peerconnection/client/peer_connection_client.h"
+
+// Forward declarations.
+typedef struct _GtkWidget GtkWidget;
+typedef union _GdkEvent GdkEvent;
+typedef struct _GdkEventKey GdkEventKey;
+typedef struct _GtkTreeView GtkTreeView;
+typedef struct _GtkTreePath GtkTreePath;
+typedef struct _GtkTreeViewColumn GtkTreeViewColumn;
+typedef struct _cairo cairo_t;
+
+// Implements the main UI of the peer connection client.
+// This is functionally equivalent to the MainWnd class in the Windows
+// implementation.
+class GtkMainWnd : public MainWindow {
+ public:
+  GtkMainWnd(const char* server, int port, bool autoconnect, bool autocall);
+  ~GtkMainWnd();
+
+  virtual void RegisterObserver(MainWndCallback* callback);
+  virtual bool IsWindow();
+  virtual void SwitchToConnectUI();
+  virtual void SwitchToPeerList(const Peers& peers);
+  virtual void SwitchToStreamingUI();
+  virtual void MessageBox(const char* caption, const char* text, bool is_error);
+  virtual MainWindow::UI current_ui();
+  virtual void StartLocalRenderer(webrtc::VideoTrackInterface* local_video);
+  virtual void StopLocalRenderer();
+  virtual void StartRemoteRenderer(webrtc::VideoTrackInterface* remote_video);
+  virtual void StopRemoteRenderer();
+
+  virtual void QueueUIThreadCallback(int msg_id, void* data);
+
+  // Creates and shows the main window with the |Connect UI| enabled.
+  bool Create();
+
+  // Destroys the window.  When the window is destroyed, it ends the
+  // main message loop.
+  bool Destroy();
+
+  // Callback for when the main window is destroyed.
+  void OnDestroyed(GtkWidget* widget, GdkEvent* event);
+
+  // Callback for when the user clicks the "Connect" button.
+  void OnClicked(GtkWidget* widget);
+
+  // Callback for keystrokes.  Used to capture Esc and Return.
+  void OnKeyPress(GtkWidget* widget, GdkEventKey* key);
+
+  // Callback when the user double clicks a peer in order to initiate a
+  // connection.
+  void OnRowActivated(GtkTreeView* tree_view,
+                      GtkTreePath* path,
+                      GtkTreeViewColumn* column);
+
+  void OnRedraw();
+
+  void Draw(GtkWidget* widget, cairo_t* cr);
+
+ protected:
+  class VideoRenderer : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+   public:
+    VideoRenderer(GtkMainWnd* main_wnd,
+                  webrtc::VideoTrackInterface* track_to_render);
+    virtual ~VideoRenderer();
+
+    // VideoSinkInterface implementation
+    void OnFrame(const webrtc::VideoFrame& frame) override;
+
+    const uint8_t* image() const { return image_.get(); }
+
+    int width() const { return width_; }
+
+    int height() const { return height_; }
+
+   protected:
+    void SetSize(int width, int height);
+    std::unique_ptr<uint8_t[]> image_;
+    int width_;
+    int height_;
+    GtkMainWnd* main_wnd_;
+    rtc::scoped_refptr<webrtc::VideoTrackInterface> rendered_track_;
+  };
+
+ protected:
+  GtkWidget* window_;     // Our main window.
+  GtkWidget* draw_area_;  // The drawing surface for rendering video streams.
+  GtkWidget* vbox_;       // Container for the Connect UI.
+  GtkWidget* server_edit_;
+  GtkWidget* port_edit_;
+  GtkWidget* peer_list_;  // The list of peers.
+  MainWndCallback* callback_;
+  std::string server_;
+  std::string port_;
+  bool autoconnect_;
+  bool autocall_;
+  std::unique_ptr<VideoRenderer> local_renderer_;
+  std::unique_ptr<VideoRenderer> remote_renderer_;
+  int width_;
+  int height_;
+  std::unique_ptr<uint8_t[]> draw_buffer_;
+  int draw_buffer_size_;
+};
+
+#endif  // EXAMPLES_PEERCONNECTION_CLIENT_LINUX_MAIN_WND_H_
diff --git a/third_party/libwebrtc/examples/peerconnection/client/main.cc b/third_party/libwebrtc/examples/peerconnection/client/main.cc
new file mode 100644
index 0000000000..32bc52bda4
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/client/main.cc
@@ -0,0 +1,133 @@
+/*
+ *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// clang-format off
+// clang formating would change include order.
+#include <windows.h>
+#include <shellapi.h>  // must come after windows.h
+// clang-format on
+
+#include <string>
+#include <vector>
+
+#include "absl/flags/parse.h"
+#include "examples/peerconnection/client/conductor.h"
+#include "examples/peerconnection/client/flag_defs.h"
+#include "examples/peerconnection/client/main_wnd.h"
+#include "examples/peerconnection/client/peer_connection_client.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/ssl_adapter.h"
+#include "rtc_base/string_utils.h"  // For ToUtf8
+#include "rtc_base/win32_socket_init.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/field_trial.h"
+
+namespace {
+// A helper class to translate Windows command line arguments into UTF8,
+// which then allows us to just pass them to the flags system.
+// This encapsulates all the work of getting the command line and translating
+// it to an array of 8-bit strings; all you have to do is create one of these,
+// and then call argc() and argv().
+class WindowsCommandLineArguments {
+ public:
+  WindowsCommandLineArguments();
+
+  WindowsCommandLineArguments(const WindowsCommandLineArguments&) = delete;
+  WindowsCommandLineArguments& operator=(WindowsCommandLineArguments&) = delete;
+
+  int argc() { return argv_.size(); }
+  char** argv() { return argv_.data(); }
+
+ private:
+  // Owned argument strings.
+  std::vector<std::string> args_;
+  // Pointers, to get layout compatible with char** argv.
+  std::vector<char*> argv_;
+};
+
+WindowsCommandLineArguments::WindowsCommandLineArguments() {
+  // start by getting the command line.
+  LPCWSTR command_line = ::GetCommandLineW();
+  // now, convert it to a list of wide char strings.
+  int argc;
+  LPWSTR* wide_argv = ::CommandLineToArgvW(command_line, &argc);
+
+  // iterate over the returned wide strings;
+  for (int i = 0; i < argc; ++i) {
+    args_.push_back(rtc::ToUtf8(wide_argv[i], wcslen(wide_argv[i])));
+    // make sure the argv array points to the string data.
+    argv_.push_back(const_cast<char*>(args_.back().c_str()));
+  }
+  LocalFree(wide_argv);
+}
+
+}  // namespace
+int PASCAL wWinMain(HINSTANCE instance,
+                    HINSTANCE prev_instance,
+                    wchar_t* cmd_line,
+                    int cmd_show) {
+  rtc::WinsockInitializer winsock_init;
+  rtc::PhysicalSocketServer ss;
+  rtc::AutoSocketServerThread main_thread(&ss);
+
+  WindowsCommandLineArguments win_args;
+  int argc = win_args.argc();
+  char** argv = win_args.argv();
+
+  absl::ParseCommandLine(argc, argv);
+
+  // InitFieldTrialsFromString stores the char*, so the char array must outlive
+  // the application.
+  const std::string forced_field_trials =
+      absl::GetFlag(FLAGS_force_fieldtrials);
+  webrtc::field_trial::InitFieldTrialsFromString(forced_field_trials.c_str());
+
+  // Abort if the user specifies a port that is outside the allowed
+  // range [1, 65535].
+  if ((absl::GetFlag(FLAGS_port) < 1) || (absl::GetFlag(FLAGS_port) > 65535)) {
+    printf("Error: %i is not a valid port.\n", absl::GetFlag(FLAGS_port));
+    return -1;
+  }
+
+  const std::string server = absl::GetFlag(FLAGS_server);
+  MainWnd wnd(server.c_str(), absl::GetFlag(FLAGS_port),
+              absl::GetFlag(FLAGS_autoconnect), absl::GetFlag(FLAGS_autocall));
+  if (!wnd.Create()) {
+    RTC_DCHECK_NOTREACHED();
+    return -1;
+  }
+
+  rtc::InitializeSSL();
+  PeerConnectionClient client;
+  auto conductor = rtc::make_ref_counted<Conductor>(&client, &wnd);
+
+  // Main loop.
+  MSG msg;
+  BOOL gm;
+  while ((gm = ::GetMessage(&msg, NULL, 0, 0)) != 0 && gm != -1) {
+    if (!wnd.PreTranslateMessage(&msg)) {
+      ::TranslateMessage(&msg);
+      ::DispatchMessage(&msg);
+    }
+  }
+
+  if (conductor->connection_active() || client.is_connected()) {
+    while ((conductor->connection_active() || client.is_connected()) &&
+           (gm = ::GetMessage(&msg, NULL, 0, 0)) != 0 && gm != -1) {
+      if (!wnd.PreTranslateMessage(&msg)) {
+        ::TranslateMessage(&msg);
+        ::DispatchMessage(&msg);
+      }
+    }
+  }
+
+  rtc::CleanupSSL();
+  return 0;
+}
diff --git a/third_party/libwebrtc/examples/peerconnection/client/main_wnd.cc b/third_party/libwebrtc/examples/peerconnection/client/main_wnd.cc
new file mode 100644
index 0000000000..afafa621b3
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/client/main_wnd.cc
@@ -0,0 +1,633 @@
+/*
+ *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/peerconnection/client/main_wnd.h"
+
+#include <math.h>
+
+#include "api/video/i420_buffer.h"
+#include "examples/peerconnection/client/defaults.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "third_party/libyuv/include/libyuv/convert_argb.h"
+
+ATOM MainWnd::wnd_class_ = 0;
+const wchar_t MainWnd::kClassName[] = L"WebRTC_MainWnd";
+
+namespace {
+
+const char kConnecting[] = "Connecting... ";
+const char kNoVideoStreams[] = "(no video streams either way)";
+const char kNoIncomingStream[] = "(no incoming video)";
+
+void CalculateWindowSizeForText(HWND wnd,
+                                const wchar_t* text,
+                                size_t* width,
+                                size_t* height) {
+  HDC dc = ::GetDC(wnd);
+  RECT text_rc = {0};
+  ::DrawTextW(dc, text, -1, &text_rc, DT_CALCRECT | DT_SINGLELINE);
+  ::ReleaseDC(wnd, dc);
+  RECT client, window;
+  ::GetClientRect(wnd, &client);
+  ::GetWindowRect(wnd, &window);
+
+  *width = text_rc.right - text_rc.left;
+  *width += (window.right - window.left) - (client.right - client.left);
+  *height = text_rc.bottom - text_rc.top;
+  *height += (window.bottom - window.top) - (client.bottom - client.top);
+}
+
+HFONT GetDefaultFont() {
+  static HFONT font = reinterpret_cast<HFONT>(GetStockObject(DEFAULT_GUI_FONT));
+  return font;
+}
+
+std::string GetWindowText(HWND wnd) {
+  char text[MAX_PATH] = {0};
+  ::GetWindowTextA(wnd, &text[0], ARRAYSIZE(text));
+  return text;
+}
+
+void AddListBoxItem(HWND listbox, const std::string& str, LPARAM item_data) {
+  LRESULT index = ::SendMessageA(listbox, LB_ADDSTRING, 0,
+                                 reinterpret_cast<LPARAM>(str.c_str()));
+  ::SendMessageA(listbox, LB_SETITEMDATA, index, item_data);
+}
+
+}  // namespace
+
+MainWnd::MainWnd(const char* server,
+                 int port,
+                 bool auto_connect,
+                 bool auto_call)
+    : ui_(CONNECT_TO_SERVER),
+      wnd_(NULL),
+      edit1_(NULL),
+      edit2_(NULL),
+      label1_(NULL),
+      label2_(NULL),
+      button_(NULL),
+      listbox_(NULL),
+      destroyed_(false),
+      nested_msg_(NULL),
+      callback_(NULL),
+      server_(server),
+      auto_connect_(auto_connect),
+      auto_call_(auto_call) {
+  char buffer[10];
+  snprintf(buffer, sizeof(buffer), "%i", port);
+  port_ = buffer;
+}
+
+MainWnd::~MainWnd() {
+  RTC_DCHECK(!IsWindow());
+}
+
+bool MainWnd::Create() {
+  RTC_DCHECK(wnd_ == NULL);
+  if (!RegisterWindowClass())
+    return false;
+
+  ui_thread_id_ = ::GetCurrentThreadId();
+  wnd_ =
+      ::CreateWindowExW(WS_EX_OVERLAPPEDWINDOW, kClassName, L"WebRTC",
+                        WS_OVERLAPPEDWINDOW | WS_VISIBLE | WS_CLIPCHILDREN,
+                        CW_USEDEFAULT, CW_USEDEFAULT, CW_USEDEFAULT,
+                        CW_USEDEFAULT, NULL, NULL, GetModuleHandle(NULL), this);
+
+  ::SendMessage(wnd_, WM_SETFONT, reinterpret_cast<WPARAM>(GetDefaultFont()),
+                TRUE);
+
+  CreateChildWindows();
+  SwitchToConnectUI();
+
+  return wnd_ != NULL;
+}
+
+bool MainWnd::Destroy() {
+  BOOL ret = FALSE;
+  if (IsWindow()) {
+    ret = ::DestroyWindow(wnd_);
+  }
+
+  return ret != FALSE;
+}
+
+void MainWnd::RegisterObserver(MainWndCallback* callback) {
+  callback_ = callback;
+}
+
+bool MainWnd::IsWindow() {
+  return wnd_ && ::IsWindow(wnd_) != FALSE;
+}
+
+bool MainWnd::PreTranslateMessage(MSG* msg) {
+  bool ret = false;
+  if (msg->message == WM_CHAR) {
+    if (msg->wParam == VK_TAB) {
+      HandleTabbing();
+      ret = true;
+    } else if (msg->wParam == VK_RETURN) {
+      OnDefaultAction();
+      ret = true;
+    } else if (msg->wParam == VK_ESCAPE) {
+      if (callback_) {
+        if (ui_ == STREAMING) {
+          callback_->DisconnectFromCurrentPeer();
+        } else {
+          callback_->DisconnectFromServer();
+        }
+      }
+    }
+  } else if (msg->hwnd == NULL && msg->message == UI_THREAD_CALLBACK) {
+    callback_->UIThreadCallback(static_cast<int>(msg->wParam),
+                                reinterpret_cast<void*>(msg->lParam));
+    ret = true;
+  }
+  return ret;
+}
+
+void MainWnd::SwitchToConnectUI() {
+  RTC_DCHECK(IsWindow());
+  LayoutPeerListUI(false);
+  ui_ = CONNECT_TO_SERVER;
+  LayoutConnectUI(true);
+  ::SetFocus(edit1_);
+
+  if (auto_connect_)
+    ::PostMessage(button_, BM_CLICK, 0, 0);
+}
+
+void MainWnd::SwitchToPeerList(const Peers& peers) {
+  LayoutConnectUI(false);
+
+  ::SendMessage(listbox_, LB_RESETCONTENT, 0, 0);
+
+  AddListBoxItem(listbox_, "List of currently connected peers:", -1);
+  Peers::const_iterator i = peers.begin();
+  for (; i != peers.end(); ++i)
+    AddListBoxItem(listbox_, i->second.c_str(), i->first);
+
+  ui_ = LIST_PEERS;
+  LayoutPeerListUI(true);
+  ::SetFocus(listbox_);
+
+  if (auto_call_ && peers.begin() != peers.end()) {
+    // Get the number of items in the list
+    LRESULT count = ::SendMessage(listbox_, LB_GETCOUNT, 0, 0);
+    if (count != LB_ERR) {
+      // Select the last item in the list
+      LRESULT selection = ::SendMessage(listbox_, LB_SETCURSEL, count - 1, 0);
+      if (selection != LB_ERR)
+        ::PostMessage(wnd_, WM_COMMAND,
+                      MAKEWPARAM(GetDlgCtrlID(listbox_), LBN_DBLCLK),
+                      reinterpret_cast<LPARAM>(listbox_));
+    }
+  }
+}
+
+void MainWnd::SwitchToStreamingUI() {
+  LayoutConnectUI(false);
+  LayoutPeerListUI(false);
+  ui_ = STREAMING;
+}
+
+void MainWnd::MessageBox(const char* caption, const char* text, bool is_error) {
+  DWORD flags = MB_OK;
+  if (is_error)
+    flags |= MB_ICONERROR;
+
+  ::MessageBoxA(handle(), text, caption, flags);
+}
+
+void MainWnd::StartLocalRenderer(webrtc::VideoTrackInterface* local_video) {
+  local_renderer_.reset(new VideoRenderer(handle(), 1, 1, local_video));
+}
+
+void MainWnd::StopLocalRenderer() {
+  local_renderer_.reset();
+}
+
+void MainWnd::StartRemoteRenderer(webrtc::VideoTrackInterface* remote_video) {
+  remote_renderer_.reset(new VideoRenderer(handle(), 1, 1, remote_video));
+}
+
+void MainWnd::StopRemoteRenderer() {
+  remote_renderer_.reset();
+}
+
+void MainWnd::QueueUIThreadCallback(int msg_id, void* data) {
+  ::PostThreadMessage(ui_thread_id_, UI_THREAD_CALLBACK,
+                      static_cast<WPARAM>(msg_id),
+                      reinterpret_cast<LPARAM>(data));
+}
+
+void MainWnd::OnPaint() {
+  PAINTSTRUCT ps;
+  ::BeginPaint(handle(), &ps);
+
+  RECT rc;
+  ::GetClientRect(handle(), &rc);
+
+  VideoRenderer* local_renderer = local_renderer_.get();
+  VideoRenderer* remote_renderer = remote_renderer_.get();
+  if (ui_ == STREAMING && remote_renderer && local_renderer) {
+    AutoLock<VideoRenderer> local_lock(local_renderer);
+    AutoLock<VideoRenderer> remote_lock(remote_renderer);
+
+    const BITMAPINFO& bmi = remote_renderer->bmi();
+    int height = abs(bmi.bmiHeader.biHeight);
+    int width = bmi.bmiHeader.biWidth;
+
+    const uint8_t* image = remote_renderer->image();
+    if (image != NULL) {
+      HDC dc_mem = ::CreateCompatibleDC(ps.hdc);
+      ::SetStretchBltMode(dc_mem, HALFTONE);
+
+      // Set the map mode so that the ratio will be maintained for us.
+      HDC all_dc[] = {ps.hdc, dc_mem};
+      for (size_t i = 0; i < arraysize(all_dc); ++i) {
+        SetMapMode(all_dc[i], MM_ISOTROPIC);
+        SetWindowExtEx(all_dc[i], width, height, NULL);
+        SetViewportExtEx(all_dc[i], rc.right, rc.bottom, NULL);
+      }
+
+      HBITMAP bmp_mem = ::CreateCompatibleBitmap(ps.hdc, rc.right, rc.bottom);
+      HGDIOBJ bmp_old = ::SelectObject(dc_mem, bmp_mem);
+
+      POINT logical_area = {rc.right, rc.bottom};
+      DPtoLP(ps.hdc, &logical_area, 1);
+
+      HBRUSH brush = ::CreateSolidBrush(RGB(0, 0, 0));
+      RECT logical_rect = {0, 0, logical_area.x, logical_area.y};
+      ::FillRect(dc_mem, &logical_rect, brush);
+      ::DeleteObject(brush);
+
+      int x = (logical_area.x / 2) - (width / 2);
+      int y = (logical_area.y / 2) - (height / 2);
+
+      StretchDIBits(dc_mem, x, y, width, height, 0, 0, width, height, image,
+                    &bmi, DIB_RGB_COLORS, SRCCOPY);
+
+      if ((rc.right - rc.left) > 200 && (rc.bottom - rc.top) > 200) {
+        const BITMAPINFO& bmi = local_renderer->bmi();
+        image = local_renderer->image();
+        int thumb_width = bmi.bmiHeader.biWidth / 4;
+        int thumb_height = abs(bmi.bmiHeader.biHeight) / 4;
+        StretchDIBits(dc_mem, logical_area.x - thumb_width - 10,
+                      logical_area.y - thumb_height - 10, thumb_width,
+                      thumb_height, 0, 0, bmi.bmiHeader.biWidth,
+                      -bmi.bmiHeader.biHeight, image, &bmi, DIB_RGB_COLORS,
+                      SRCCOPY);
+      }
+
+      BitBlt(ps.hdc, 0, 0, logical_area.x, logical_area.y, dc_mem, 0, 0,
+             SRCCOPY);
+
+      // Cleanup.
+      ::SelectObject(dc_mem, bmp_old);
+      ::DeleteObject(bmp_mem);
+      ::DeleteDC(dc_mem);
+    } else {
+      // We're still waiting for the video stream to be initialized.
+      HBRUSH brush = ::CreateSolidBrush(RGB(0, 0, 0));
+      ::FillRect(ps.hdc, &rc, brush);
+      ::DeleteObject(brush);
+
+      HGDIOBJ old_font = ::SelectObject(ps.hdc, GetDefaultFont());
+      ::SetTextColor(ps.hdc, RGB(0xff, 0xff, 0xff));
+      ::SetBkMode(ps.hdc, TRANSPARENT);
+
+      std::string text(kConnecting);
+      if (!local_renderer->image()) {
+        text += kNoVideoStreams;
+      } else {
+        text += kNoIncomingStream;
+      }
+      ::DrawTextA(ps.hdc, text.c_str(), -1, &rc,
+                  DT_SINGLELINE | DT_CENTER | DT_VCENTER);
+      ::SelectObject(ps.hdc, old_font);
+    }
+  } else {
+    HBRUSH brush = ::CreateSolidBrush(::GetSysColor(COLOR_WINDOW));
+    ::FillRect(ps.hdc, &rc, brush);
+    ::DeleteObject(brush);
+  }
+
+  ::EndPaint(handle(), &ps);
+}
+
+void MainWnd::OnDestroyed() {
+  PostQuitMessage(0);
+}
+
+void MainWnd::OnDefaultAction() {
+  if (!callback_)
+    return;
+  if (ui_ == CONNECT_TO_SERVER) {
+    std::string server(GetWindowText(edit1_));
+    std::string port_str(GetWindowText(edit2_));
+    int port = port_str.length() ? atoi(port_str.c_str()) : 0;
+    callback_->StartLogin(server, port);
+  } else if (ui_ == LIST_PEERS) {
+    LRESULT sel = ::SendMessage(listbox_, LB_GETCURSEL, 0, 0);
+    if (sel != LB_ERR) {
+      LRESULT peer_id = ::SendMessage(listbox_, LB_GETITEMDATA, sel, 0);
+      if (peer_id != -1 && callback_) {
+        callback_->ConnectToPeer(peer_id);
+      }
+    }
+  } else {
+    ::MessageBoxA(wnd_, "OK!", "Yeah", MB_OK);
+  }
+}
+
+bool MainWnd::OnMessage(UINT msg, WPARAM wp, LPARAM lp, LRESULT* result) {
+  switch (msg) {
+    case WM_ERASEBKGND:
+      *result = TRUE;
+      return true;
+
+    case WM_PAINT:
+      OnPaint();
+      return true;
+
+    case WM_SETFOCUS:
+      if (ui_ == CONNECT_TO_SERVER) {
+        SetFocus(edit1_);
+      } else if (ui_ == LIST_PEERS) {
+        SetFocus(listbox_);
+      }
+      return true;
+
+    case WM_SIZE:
+      if (ui_ == CONNECT_TO_SERVER) {
+        LayoutConnectUI(true);
+      } else if (ui_ == LIST_PEERS) {
+        LayoutPeerListUI(true);
+      }
+      break;
+
+    case WM_CTLCOLORSTATIC:
+      *result = reinterpret_cast<LRESULT>(GetSysColorBrush(COLOR_WINDOW));
+      return true;
+
+    case WM_COMMAND:
+      if (button_ == reinterpret_cast<HWND>(lp)) {
+        if (BN_CLICKED == HIWORD(wp))
+          OnDefaultAction();
+      } else if (listbox_ == reinterpret_cast<HWND>(lp)) {
+        if (LBN_DBLCLK == HIWORD(wp)) {
+          OnDefaultAction();
+        }
+      }
+      return true;
+
+    case WM_CLOSE:
+      if (callback_)
+        callback_->Close();
+      break;
+  }
+  return false;
+}
+
+// static
+LRESULT CALLBACK MainWnd::WndProc(HWND hwnd, UINT msg, WPARAM wp, LPARAM lp) {
+  MainWnd* me =
+      reinterpret_cast<MainWnd*>(::GetWindowLongPtr(hwnd, GWLP_USERDATA));
+  if (!me && WM_CREATE == msg) {
+    CREATESTRUCT* cs = reinterpret_cast<CREATESTRUCT*>(lp);
+    me = reinterpret_cast<MainWnd*>(cs->lpCreateParams);
+    me->wnd_ = hwnd;
+    ::SetWindowLongPtr(hwnd, GWLP_USERDATA, reinterpret_cast<LONG_PTR>(me));
+  }
+
+  LRESULT result = 0;
+  if (me) {
+    void* prev_nested_msg = me->nested_msg_;
+    me->nested_msg_ = &msg;
+
+    bool handled = me->OnMessage(msg, wp, lp, &result);
+    if (WM_NCDESTROY == msg) {
+      me->destroyed_ = true;
+    } else if (!handled) {
+      result = ::DefWindowProc(hwnd, msg, wp, lp);
+    }
+
+    if (me->destroyed_ && prev_nested_msg == NULL) {
+      me->OnDestroyed();
+      me->wnd_ = NULL;
+      me->destroyed_ = false;
+    }
+
+    me->nested_msg_ = prev_nested_msg;
+  } else {
+    result = ::DefWindowProc(hwnd, msg, wp, lp);
+  }
+
+  return result;
+}
+
+// static
+bool MainWnd::RegisterWindowClass() {
+  if (wnd_class_)
+    return true;
+
+  WNDCLASSEXW wcex = {sizeof(WNDCLASSEX)};
+  wcex.style = CS_DBLCLKS;
+  wcex.hInstance = GetModuleHandle(NULL);
+  wcex.hbrBackground = reinterpret_cast<HBRUSH>(COLOR_WINDOW + 1);
+  wcex.hCursor = ::LoadCursor(NULL, IDC_ARROW);
+  wcex.lpfnWndProc = &WndProc;
+  wcex.lpszClassName = kClassName;
+  wnd_class_ = ::RegisterClassExW(&wcex);
+  RTC_DCHECK(wnd_class_ != 0);
+  return wnd_class_ != 0;
+}
+
+void MainWnd::CreateChildWindow(HWND* wnd,
+                                MainWnd::ChildWindowID id,
+                                const wchar_t* class_name,
+                                DWORD control_style,
+                                DWORD ex_style) {
+  if (::IsWindow(*wnd))
+    return;
+
+  // Child windows are invisible at first, and shown after being resized.
+  DWORD style = WS_CHILD | control_style;
+  *wnd = ::CreateWindowExW(ex_style, class_name, L"", style, 100, 100, 100, 100,
+                           wnd_, reinterpret_cast<HMENU>(id),
+                           GetModuleHandle(NULL), NULL);
+  RTC_DCHECK(::IsWindow(*wnd) != FALSE);
+  ::SendMessage(*wnd, WM_SETFONT, reinterpret_cast<WPARAM>(GetDefaultFont()),
+                TRUE);
+}
+
+void MainWnd::CreateChildWindows() {
+  // Create the child windows in tab order.
+  CreateChildWindow(&label1_, LABEL1_ID, L"Static", ES_CENTER | ES_READONLY, 0);
+  CreateChildWindow(&edit1_, EDIT_ID, L"Edit",
+                    ES_LEFT | ES_NOHIDESEL | WS_TABSTOP, WS_EX_CLIENTEDGE);
+  CreateChildWindow(&label2_, LABEL2_ID, L"Static", ES_CENTER | ES_READONLY, 0);
+  CreateChildWindow(&edit2_, EDIT_ID, L"Edit",
+                    ES_LEFT | ES_NOHIDESEL | WS_TABSTOP, WS_EX_CLIENTEDGE);
+  CreateChildWindow(&button_, BUTTON_ID, L"Button", BS_CENTER | WS_TABSTOP, 0);
+
+  CreateChildWindow(&listbox_, LISTBOX_ID, L"ListBox",
+                    LBS_HASSTRINGS | LBS_NOTIFY, WS_EX_CLIENTEDGE);
+
+  ::SetWindowTextA(edit1_, server_.c_str());
+  ::SetWindowTextA(edit2_, port_.c_str());
+}
+
+void MainWnd::LayoutConnectUI(bool show) {
+  struct Windows {
+    HWND wnd;
+    const wchar_t* text;
+    size_t width;
+    size_t height;
+  } windows[] = {
+      {label1_, L"Server"},  {edit1_, L"XXXyyyYYYgggXXXyyyYYYggg"},
+      {label2_, L":"},       {edit2_, L"XyXyX"},
+      {button_, L"Connect"},
+  };
+
+  if (show) {
+    const size_t kSeparator = 5;
+    size_t total_width = (ARRAYSIZE(windows) - 1) * kSeparator;
+
+    for (size_t i = 0; i < ARRAYSIZE(windows); ++i) {
+      CalculateWindowSizeForText(windows[i].wnd, windows[i].text,
+                                 &windows[i].width, &windows[i].height);
+      total_width += windows[i].width;
+    }
+
+    RECT rc;
+    ::GetClientRect(wnd_, &rc);
+    size_t x = (rc.right / 2) - (total_width / 2);
+    size_t y = rc.bottom / 2;
+    for (size_t i = 0; i < ARRAYSIZE(windows); ++i) {
+      size_t top = y - (windows[i].height / 2);
+      ::MoveWindow(windows[i].wnd, static_cast<int>(x), static_cast<int>(top),
+                   static_cast<int>(windows[i].width),
+                   static_cast<int>(windows[i].height), TRUE);
+      x += kSeparator + windows[i].width;
+      if (windows[i].text[0] != 'X')
+        ::SetWindowTextW(windows[i].wnd, windows[i].text);
+      ::ShowWindow(windows[i].wnd, SW_SHOWNA);
+    }
+  } else {
+    for (size_t i = 0; i < ARRAYSIZE(windows); ++i) {
+      ::ShowWindow(windows[i].wnd, SW_HIDE);
+    }
+  }
+}
+
+void MainWnd::LayoutPeerListUI(bool show) {
+  if (show) {
+    RECT rc;
+    ::GetClientRect(wnd_, &rc);
+    ::MoveWindow(listbox_, 0, 0, rc.right, rc.bottom, TRUE);
+    ::ShowWindow(listbox_, SW_SHOWNA);
+  } else {
+    ::ShowWindow(listbox_, SW_HIDE);
+    InvalidateRect(wnd_, NULL, TRUE);
+  }
+}
+
+void MainWnd::HandleTabbing() {
+  bool shift = ((::GetAsyncKeyState(VK_SHIFT) & 0x8000) != 0);
+  UINT next_cmd = shift ? GW_HWNDPREV : GW_HWNDNEXT;
+  UINT loop_around_cmd = shift ? GW_HWNDLAST : GW_HWNDFIRST;
+  HWND focus = GetFocus(), next;
+  do {
+    next = ::GetWindow(focus, next_cmd);
+    if (IsWindowVisible(next) &&
+        (GetWindowLong(next, GWL_STYLE) & WS_TABSTOP)) {
+      break;
+    }
+
+    if (!next) {
+      next = ::GetWindow(focus, loop_around_cmd);
+      if (IsWindowVisible(next) &&
+          (GetWindowLong(next, GWL_STYLE) & WS_TABSTOP)) {
+        break;
+      }
+    }
+    focus = next;
+  } while (true);
+  ::SetFocus(next);
+}
+
+//
+// MainWnd::VideoRenderer
+//
+
+MainWnd::VideoRenderer::VideoRenderer(
+    HWND wnd,
+    int width,
+    int height,
+    webrtc::VideoTrackInterface* track_to_render)
+    : wnd_(wnd), rendered_track_(track_to_render) {
+  ::InitializeCriticalSection(&buffer_lock_);
+  ZeroMemory(&bmi_, sizeof(bmi_));
+  bmi_.bmiHeader.biSize = sizeof(BITMAPINFOHEADER);
+  bmi_.bmiHeader.biPlanes = 1;
+  bmi_.bmiHeader.biBitCount = 32;
+  bmi_.bmiHeader.biCompression = BI_RGB;
+  bmi_.bmiHeader.biWidth = width;
+  bmi_.bmiHeader.biHeight = -height;
+  bmi_.bmiHeader.biSizeImage =
+      width * height * (bmi_.bmiHeader.biBitCount >> 3);
+  rendered_track_->AddOrUpdateSink(this, rtc::VideoSinkWants());
+}
+
+MainWnd::VideoRenderer::~VideoRenderer() {
+  rendered_track_->RemoveSink(this);
+  ::DeleteCriticalSection(&buffer_lock_);
+}
+
+void MainWnd::VideoRenderer::SetSize(int width, int height) {
+  AutoLock<VideoRenderer> lock(this);
+
+  if (width == bmi_.bmiHeader.biWidth && height == bmi_.bmiHeader.biHeight) {
+    return;
+  }
+
+  bmi_.bmiHeader.biWidth = width;
+  bmi_.bmiHeader.biHeight = -height;
+  bmi_.bmiHeader.biSizeImage =
+      width * height * (bmi_.bmiHeader.biBitCount >> 3);
+  image_.reset(new uint8_t[bmi_.bmiHeader.biSizeImage]);
+}
+
+void MainWnd::VideoRenderer::OnFrame(const webrtc::VideoFrame& video_frame) {
+  {
+    AutoLock<VideoRenderer> lock(this);
+
+    rtc::scoped_refptr<webrtc::I420BufferInterface> buffer(
+        video_frame.video_frame_buffer()->ToI420());
+    if (video_frame.rotation() != webrtc::kVideoRotation_0) {
+      buffer = webrtc::I420Buffer::Rotate(*buffer, video_frame.rotation());
+    }
+
+    SetSize(buffer->width(), buffer->height());
+
+    RTC_DCHECK(image_.get() != NULL);
+    libyuv::I420ToARGB(buffer->DataY(), buffer->StrideY(), buffer->DataU(),
+                       buffer->StrideU(), buffer->DataV(), buffer->StrideV(),
+                       image_.get(),
+                       bmi_.bmiHeader.biWidth * bmi_.bmiHeader.biBitCount / 8,
+                       buffer->width(), buffer->height());
+  }
+  InvalidateRect(wnd_, NULL, TRUE);
+}
diff --git a/third_party/libwebrtc/examples/peerconnection/client/main_wnd.h b/third_party/libwebrtc/examples/peerconnection/client/main_wnd.h
new file mode 100644
index 0000000000..898fea9d92
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/client/main_wnd.h
@@ -0,0 +1,206 @@
+/*
+ *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_PEERCONNECTION_CLIENT_MAIN_WND_H_
+#define EXAMPLES_PEERCONNECTION_CLIENT_MAIN_WND_H_
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "api/media_stream_interface.h"
+#include "api/video/video_frame.h"
+#include "examples/peerconnection/client/peer_connection_client.h"
+#include "media/base/media_channel.h"
+#include "media/base/video_common.h"
+#if defined(WEBRTC_WIN)
+#include "rtc_base/win32.h"
+#endif  // WEBRTC_WIN
+
+class MainWndCallback {
+ public:
+  virtual void StartLogin(const std::string& server, int port) = 0;
+  virtual void DisconnectFromServer() = 0;
+  virtual void ConnectToPeer(int peer_id) = 0;
+  virtual void DisconnectFromCurrentPeer() = 0;
+  virtual void UIThreadCallback(int msg_id, void* data) = 0;
+  virtual void Close() = 0;
+
+ protected:
+  virtual ~MainWndCallback() {}
+};
+
+// Pure virtual interface for the main window.
+class MainWindow {
+ public:
+  virtual ~MainWindow() {}
+
+  enum UI {
+    CONNECT_TO_SERVER,
+    LIST_PEERS,
+    STREAMING,
+  };
+
+  virtual void RegisterObserver(MainWndCallback* callback) = 0;
+
+  virtual bool IsWindow() = 0;
+  virtual void MessageBox(const char* caption,
+                          const char* text,
+                          bool is_error) = 0;
+
+  virtual UI current_ui() = 0;
+
+  virtual void SwitchToConnectUI() = 0;
+  virtual void SwitchToPeerList(const Peers& peers) = 0;
+  virtual void SwitchToStreamingUI() = 0;
+
+  virtual void StartLocalRenderer(webrtc::VideoTrackInterface* local_video) = 0;
+  virtual void StopLocalRenderer() = 0;
+  virtual void StartRemoteRenderer(
+      webrtc::VideoTrackInterface* remote_video) = 0;
+  virtual void StopRemoteRenderer() = 0;
+
+  virtual void QueueUIThreadCallback(int msg_id, void* data) = 0;
+};
+
+#ifdef WIN32
+
+class MainWnd : public MainWindow {
+ public:
+  static const wchar_t kClassName[];
+
+  enum WindowMessages {
+    UI_THREAD_CALLBACK = WM_APP + 1,
+  };
+
+  MainWnd(const char* server, int port, bool auto_connect, bool auto_call);
+  ~MainWnd();
+
+  bool Create();
+  bool Destroy();
+  bool PreTranslateMessage(MSG* msg);
+
+  virtual void RegisterObserver(MainWndCallback* callback);
+  virtual bool IsWindow();
+  virtual void SwitchToConnectUI();
+  virtual void SwitchToPeerList(const Peers& peers);
+  virtual void SwitchToStreamingUI();
+  virtual void MessageBox(const char* caption, const char* text, bool is_error);
+  virtual UI current_ui() { return ui_; }
+
+  virtual void StartLocalRenderer(webrtc::VideoTrackInterface* local_video);
+  virtual void StopLocalRenderer();
+  virtual void StartRemoteRenderer(webrtc::VideoTrackInterface* remote_video);
+  virtual void StopRemoteRenderer();
+
+  virtual void QueueUIThreadCallback(int msg_id, void* data);
+
+  HWND handle() const { return wnd_; }
+
+  class VideoRenderer : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+   public:
+    VideoRenderer(HWND wnd,
+                  int width,
+                  int height,
+                  webrtc::VideoTrackInterface* track_to_render);
+    virtual ~VideoRenderer();
+
+    void Lock() { ::EnterCriticalSection(&buffer_lock_); }
+
+    void Unlock() { ::LeaveCriticalSection(&buffer_lock_); }
+
+    // VideoSinkInterface implementation
+    void OnFrame(const webrtc::VideoFrame& frame) override;
+
+    const BITMAPINFO& bmi() const { return bmi_; }
+    const uint8_t* image() const { return image_.get(); }
+
+   protected:
+    void SetSize(int width, int height);
+
+    enum {
+      SET_SIZE,
+      RENDER_FRAME,
+    };
+
+    HWND wnd_;
+    BITMAPINFO bmi_;
+    std::unique_ptr<uint8_t[]> image_;
+    CRITICAL_SECTION buffer_lock_;
+    rtc::scoped_refptr<webrtc::VideoTrackInterface> rendered_track_;
+  };
+
+  // A little helper class to make sure we always to proper locking and
+  // unlocking when working with VideoRenderer buffers.
+  template <typename T>
+  class AutoLock {
+   public:
+    explicit AutoLock(T* obj) : obj_(obj) { obj_->Lock(); }
+    ~AutoLock() { obj_->Unlock(); }
+
+   protected:
+    T* obj_;
+  };
+
+ protected:
+  enum ChildWindowID {
+    EDIT_ID = 1,
+    BUTTON_ID,
+    LABEL1_ID,
+    LABEL2_ID,
+    LISTBOX_ID,
+  };
+
+  void OnPaint();
+  void OnDestroyed();
+
+  void OnDefaultAction();
+
+  bool OnMessage(UINT msg, WPARAM wp, LPARAM lp, LRESULT* result);
+
+  static LRESULT CALLBACK WndProc(HWND hwnd, UINT msg, WPARAM wp, LPARAM lp);
+  static bool RegisterWindowClass();
+
+  void CreateChildWindow(HWND* wnd,
+                         ChildWindowID id,
+                         const wchar_t* class_name,
+                         DWORD control_style,
+                         DWORD ex_style);
+  void CreateChildWindows();
+
+  void LayoutConnectUI(bool show);
+  void LayoutPeerListUI(bool show);
+
+  void HandleTabbing();
+
+ private:
+  std::unique_ptr<VideoRenderer> local_renderer_;
+  std::unique_ptr<VideoRenderer> remote_renderer_;
+  UI ui_;
+  HWND wnd_;
+  DWORD ui_thread_id_;
+  HWND edit1_;
+  HWND edit2_;
+  HWND label1_;
+  HWND label2_;
+  HWND button_;
+  HWND listbox_;
+  bool destroyed_;
+  void* nested_msg_;
+  MainWndCallback* callback_;
+  static ATOM wnd_class_;
+  std::string server_;
+  std::string port_;
+  bool auto_connect_;
+  bool auto_call_;
+};
+#endif  // WIN32
+
+#endif  // EXAMPLES_PEERCONNECTION_CLIENT_MAIN_WND_H_
diff --git a/third_party/libwebrtc/examples/peerconnection/client/peer_connection_client.cc b/third_party/libwebrtc/examples/peerconnection/client/peer_connection_client.cc
new file mode 100644
index 0000000000..2746752d80
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/client/peer_connection_client.cc
@@ -0,0 +1,489 @@
+/*
+ *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/peerconnection/client/peer_connection_client.h"
+
+#include "api/units/time_delta.h"
+#include "examples/peerconnection/client/defaults.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/net_helpers.h"
+
+namespace {
+
+// This is our magical hangup signal.
+constexpr char kByeMessage[] = "BYE";
+// Delay between server connection retries, in milliseconds
+constexpr webrtc::TimeDelta kReconnectDelay = webrtc::TimeDelta::Seconds(2);
+
+rtc::Socket* CreateClientSocket(int family) {
+  rtc::Thread* thread = rtc::Thread::Current();
+  RTC_DCHECK(thread != NULL);
+  return thread->socketserver()->CreateSocket(family, SOCK_STREAM);
+}
+
+}  // namespace
+
+PeerConnectionClient::PeerConnectionClient()
+    : callback_(NULL), resolver_(NULL), state_(NOT_CONNECTED), my_id_(-1) {}
+
+PeerConnectionClient::~PeerConnectionClient() = default;
+
+void PeerConnectionClient::InitSocketSignals() {
+  RTC_DCHECK(control_socket_.get() != NULL);
+  RTC_DCHECK(hanging_get_.get() != NULL);
+  control_socket_->SignalCloseEvent.connect(this,
+                                            &PeerConnectionClient::OnClose);
+  hanging_get_->SignalCloseEvent.connect(this, &PeerConnectionClient::OnClose);
+  control_socket_->SignalConnectEvent.connect(this,
+                                              &PeerConnectionClient::OnConnect);
+  hanging_get_->SignalConnectEvent.connect(
+      this, &PeerConnectionClient::OnHangingGetConnect);
+  control_socket_->SignalReadEvent.connect(this, &PeerConnectionClient::OnRead);
+  hanging_get_->SignalReadEvent.connect(
+      this, &PeerConnectionClient::OnHangingGetRead);
+}
+
+int PeerConnectionClient::id() const {
+  return my_id_;
+}
+
+bool PeerConnectionClient::is_connected() const {
+  return my_id_ != -1;
+}
+
+const Peers& PeerConnectionClient::peers() const {
+  return peers_;
+}
+
+void PeerConnectionClient::RegisterObserver(
+    PeerConnectionClientObserver* callback) {
+  RTC_DCHECK(!callback_);
+  callback_ = callback;
+}
+
+void PeerConnectionClient::Connect(const std::string& server,
+                                   int port,
+                                   const std::string& client_name) {
+  RTC_DCHECK(!server.empty());
+  RTC_DCHECK(!client_name.empty());
+
+  if (state_ != NOT_CONNECTED) {
+    RTC_LOG(LS_WARNING)
+        << "The client must not be connected before you can call Connect()";
+    callback_->OnServerConnectionFailure();
+    return;
+  }
+
+  if (server.empty() || client_name.empty()) {
+    callback_->OnServerConnectionFailure();
+    return;
+  }
+
+  if (port <= 0)
+    port = kDefaultServerPort;
+
+  server_address_.SetIP(server);
+  server_address_.SetPort(port);
+  client_name_ = client_name;
+
+  if (server_address_.IsUnresolvedIP()) {
+    state_ = RESOLVING;
+    resolver_ = new rtc::AsyncResolver();
+    resolver_->SignalDone.connect(this, &PeerConnectionClient::OnResolveResult);
+    resolver_->Start(server_address_);
+  } else {
+    DoConnect();
+  }
+}
+
+void PeerConnectionClient::OnResolveResult(
+    rtc::AsyncResolverInterface* resolver) {
+  if (resolver_->GetError() != 0) {
+    callback_->OnServerConnectionFailure();
+    resolver_->Destroy(false);
+    resolver_ = NULL;
+    state_ = NOT_CONNECTED;
+  } else {
+    server_address_ = resolver_->address();
+    DoConnect();
+  }
+}
+
+void PeerConnectionClient::DoConnect() {
+  control_socket_.reset(CreateClientSocket(server_address_.ipaddr().family()));
+  hanging_get_.reset(CreateClientSocket(server_address_.ipaddr().family()));
+  InitSocketSignals();
+  char buffer[1024];
+  snprintf(buffer, sizeof(buffer), "GET /sign_in?%s HTTP/1.0\r\n\r\n",
+           client_name_.c_str());
+  onconnect_data_ = buffer;
+
+  bool ret = ConnectControlSocket();
+  if (ret)
+    state_ = SIGNING_IN;
+  if (!ret) {
+    callback_->OnServerConnectionFailure();
+  }
+}
+
+bool PeerConnectionClient::SendToPeer(int peer_id, const std::string& message) {
+  if (state_ != CONNECTED)
+    return false;
+
+  RTC_DCHECK(is_connected());
+  RTC_DCHECK(control_socket_->GetState() == rtc::Socket::CS_CLOSED);
+  if (!is_connected() || peer_id == -1)
+    return false;
+
+  char headers[1024];
+  snprintf(headers, sizeof(headers),
+           "POST /message?peer_id=%i&to=%i HTTP/1.0\r\n"
+           "Content-Length: %zu\r\n"
+           "Content-Type: text/plain\r\n"
+           "\r\n",
+           my_id_, peer_id, message.length());
+  onconnect_data_ = headers;
+  onconnect_data_ += message;
+  return ConnectControlSocket();
+}
+
+bool PeerConnectionClient::SendHangUp(int peer_id) {
+  return SendToPeer(peer_id, kByeMessage);
+}
+
+bool PeerConnectionClient::IsSendingMessage() {
+  return state_ == CONNECTED &&
+         control_socket_->GetState() != rtc::Socket::CS_CLOSED;
+}
+
+bool PeerConnectionClient::SignOut() {
+  if (state_ == NOT_CONNECTED || state_ == SIGNING_OUT)
+    return true;
+
+  if (hanging_get_->GetState() != rtc::Socket::CS_CLOSED)
+    hanging_get_->Close();
+
+  if (control_socket_->GetState() == rtc::Socket::CS_CLOSED) {
+    state_ = SIGNING_OUT;
+
+    if (my_id_ != -1) {
+      char buffer[1024];
+      snprintf(buffer, sizeof(buffer),
+               "GET /sign_out?peer_id=%i HTTP/1.0\r\n\r\n", my_id_);
+      onconnect_data_ = buffer;
+      return ConnectControlSocket();
+    } else {
+      // Can occur if the app is closed before we finish connecting.
+      return true;
+    }
+  } else {
+    state_ = SIGNING_OUT_WAITING;
+  }
+
+  return true;
+}
+
+void PeerConnectionClient::Close() {
+  control_socket_->Close();
+  hanging_get_->Close();
+  onconnect_data_.clear();
+  peers_.clear();
+  if (resolver_ != NULL) {
+    resolver_->Destroy(false);
+    resolver_ = NULL;
+  }
+  my_id_ = -1;
+  state_ = NOT_CONNECTED;
+}
+
+bool PeerConnectionClient::ConnectControlSocket() {
+  RTC_DCHECK(control_socket_->GetState() == rtc::Socket::CS_CLOSED);
+  int err = control_socket_->Connect(server_address_);
+  if (err == SOCKET_ERROR) {
+    Close();
+    return false;
+  }
+  return true;
+}
+
+void PeerConnectionClient::OnConnect(rtc::Socket* socket) {
+  RTC_DCHECK(!onconnect_data_.empty());
+  size_t sent = socket->Send(onconnect_data_.c_str(), onconnect_data_.length());
+  RTC_DCHECK(sent == onconnect_data_.length());
+  onconnect_data_.clear();
+}
+
+void PeerConnectionClient::OnHangingGetConnect(rtc::Socket* socket) {
+  char buffer[1024];
+  snprintf(buffer, sizeof(buffer), "GET /wait?peer_id=%i HTTP/1.0\r\n\r\n",
+           my_id_);
+  int len = static_cast<int>(strlen(buffer));
+  int sent = socket->Send(buffer, len);
+  RTC_DCHECK(sent == len);
+}
+
+void PeerConnectionClient::OnMessageFromPeer(int peer_id,
+                                             const std::string& message) {
+  if (message.length() == (sizeof(kByeMessage) - 1) &&
+      message.compare(kByeMessage) == 0) {
+    callback_->OnPeerDisconnected(peer_id);
+  } else {
+    callback_->OnMessageFromPeer(peer_id, message);
+  }
+}
+
+bool PeerConnectionClient::GetHeaderValue(const std::string& data,
+                                          size_t eoh,
+                                          const char* header_pattern,
+                                          size_t* value) {
+  RTC_DCHECK(value != NULL);
+  size_t found = data.find(header_pattern);
+  if (found != std::string::npos && found < eoh) {
+    *value = atoi(&data[found + strlen(header_pattern)]);
+    return true;
+  }
+  return false;
+}
+
+bool PeerConnectionClient::GetHeaderValue(const std::string& data,
+                                          size_t eoh,
+                                          const char* header_pattern,
+                                          std::string* value) {
+  RTC_DCHECK(value != NULL);
+  size_t found = data.find(header_pattern);
+  if (found != std::string::npos && found < eoh) {
+    size_t begin = found + strlen(header_pattern);
+    size_t end = data.find("\r\n", begin);
+    if (end == std::string::npos)
+      end = eoh;
+    value->assign(data.substr(begin, end - begin));
+    return true;
+  }
+  return false;
+}
+
+bool PeerConnectionClient::ReadIntoBuffer(rtc::Socket* socket,
+                                          std::string* data,
+                                          size_t* content_length) {
+  char buffer[0xffff];
+  do {
+    int bytes = socket->Recv(buffer, sizeof(buffer), nullptr);
+    if (bytes <= 0)
+      break;
+    data->append(buffer, bytes);
+  } while (true);
+
+  bool ret = false;
+  size_t i = data->find("\r\n\r\n");
+  if (i != std::string::npos) {
+    RTC_LOG(LS_INFO) << "Headers received";
+    if (GetHeaderValue(*data, i, "\r\nContent-Length: ", content_length)) {
+      size_t total_response_size = (i + 4) + *content_length;
+      if (data->length() >= total_response_size) {
+        ret = true;
+        std::string should_close;
+        const char kConnection[] = "\r\nConnection: ";
+        if (GetHeaderValue(*data, i, kConnection, &should_close) &&
+            should_close.compare("close") == 0) {
+          socket->Close();
+          // Since we closed the socket, there was no notification delivered
+          // to us.  Compensate by letting ourselves know.
+          OnClose(socket, 0);
+        }
+      } else {
+        // We haven't received everything.  Just continue to accept data.
+      }
+    } else {
+      RTC_LOG(LS_ERROR) << "No content length field specified by the server.";
+    }
+  }
+  return ret;
+}
+
+void PeerConnectionClient::OnRead(rtc::Socket* socket) {
+  size_t content_length = 0;
+  if (ReadIntoBuffer(socket, &control_data_, &content_length)) {
+    size_t peer_id = 0, eoh = 0;
+    bool ok =
+        ParseServerResponse(control_data_, content_length, &peer_id, &eoh);
+    if (ok) {
+      if (my_id_ == -1) {
+        // First response.  Let's store our server assigned ID.
+        RTC_DCHECK(state_ == SIGNING_IN);
+        my_id_ = static_cast<int>(peer_id);
+        RTC_DCHECK(my_id_ != -1);
+
+        // The body of the response will be a list of already connected peers.
+        if (content_length) {
+          size_t pos = eoh + 4;
+          while (pos < control_data_.size()) {
+            size_t eol = control_data_.find('\n', pos);
+            if (eol == std::string::npos)
+              break;
+            int id = 0;
+            std::string name;
+            bool connected;
+            if (ParseEntry(control_data_.substr(pos, eol - pos), &name, &id,
+                           &connected) &&
+                id != my_id_) {
+              peers_[id] = name;
+              callback_->OnPeerConnected(id, name);
+            }
+            pos = eol + 1;
+          }
+        }
+        RTC_DCHECK(is_connected());
+        callback_->OnSignedIn();
+      } else if (state_ == SIGNING_OUT) {
+        Close();
+        callback_->OnDisconnected();
+      } else if (state_ == SIGNING_OUT_WAITING) {
+        SignOut();
+      }
+    }
+
+    control_data_.clear();
+
+    if (state_ == SIGNING_IN) {
+      RTC_DCHECK(hanging_get_->GetState() == rtc::Socket::CS_CLOSED);
+      state_ = CONNECTED;
+      hanging_get_->Connect(server_address_);
+    }
+  }
+}
+
+void PeerConnectionClient::OnHangingGetRead(rtc::Socket* socket) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  size_t content_length = 0;
+  if (ReadIntoBuffer(socket, &notification_data_, &content_length)) {
+    size_t peer_id = 0, eoh = 0;
+    bool ok =
+        ParseServerResponse(notification_data_, content_length, &peer_id, &eoh);
+
+    if (ok) {
+      // Store the position where the body begins.
+      size_t pos = eoh + 4;
+
+      if (my_id_ == static_cast<int>(peer_id)) {
+        // A notification about a new member or a member that just
+        // disconnected.
+        int id = 0;
+        std::string name;
+        bool connected = false;
+        if (ParseEntry(notification_data_.substr(pos), &name, &id,
+                       &connected)) {
+          if (connected) {
+            peers_[id] = name;
+            callback_->OnPeerConnected(id, name);
+          } else {
+            peers_.erase(id);
+            callback_->OnPeerDisconnected(id);
+          }
+        }
+      } else {
+        OnMessageFromPeer(static_cast<int>(peer_id),
+                          notification_data_.substr(pos));
+      }
+    }
+
+    notification_data_.clear();
+  }
+
+  if (hanging_get_->GetState() == rtc::Socket::CS_CLOSED &&
+      state_ == CONNECTED) {
+    hanging_get_->Connect(server_address_);
+  }
+}
+
+bool PeerConnectionClient::ParseEntry(const std::string& entry,
+                                      std::string* name,
+                                      int* id,
+                                      bool* connected) {
+  RTC_DCHECK(name != NULL);
+  RTC_DCHECK(id != NULL);
+  RTC_DCHECK(connected != NULL);
+  RTC_DCHECK(!entry.empty());
+
+  *connected = false;
+  size_t separator = entry.find(',');
+  if (separator != std::string::npos) {
+    *id = atoi(&entry[separator + 1]);
+    name->assign(entry.substr(0, separator));
+    separator = entry.find(',', separator + 1);
+    if (separator != std::string::npos) {
+      *connected = atoi(&entry[separator + 1]) ? true : false;
+    }
+  }
+  return !name->empty();
+}
+
+int PeerConnectionClient::GetResponseStatus(const std::string& response) {
+  int status = -1;
+  size_t pos = response.find(' ');
+  if (pos != std::string::npos)
+    status = atoi(&response[pos + 1]);
+  return status;
+}
+
+bool PeerConnectionClient::ParseServerResponse(const std::string& response,
+                                               size_t content_length,
+                                               size_t* peer_id,
+                                               size_t* eoh) {
+  int status = GetResponseStatus(response.c_str());
+  if (status != 200) {
+    RTC_LOG(LS_ERROR) << "Received error from server";
+    Close();
+    callback_->OnDisconnected();
+    return false;
+  }
+
+  *eoh = response.find("\r\n\r\n");
+  RTC_DCHECK(*eoh != std::string::npos);
+  if (*eoh == std::string::npos)
+    return false;
+
+  *peer_id = -1;
+
+  // See comment in peer_channel.cc for why we use the Pragma header.
+  GetHeaderValue(response, *eoh, "\r\nPragma: ", peer_id);
+
+  return true;
+}
+
+void PeerConnectionClient::OnClose(rtc::Socket* socket, int err) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+
+  socket->Close();
+
+#ifdef WIN32
+  if (err != WSAECONNREFUSED) {
+#else
+  if (err != ECONNREFUSED) {
+#endif
+    if (socket == hanging_get_.get()) {
+      if (state_ == CONNECTED) {
+        hanging_get_->Close();
+        hanging_get_->Connect(server_address_);
+      }
+    } else {
+      callback_->OnMessageSent(err);
+    }
+  } else {
+    if (socket == control_socket_.get()) {
+      RTC_LOG(LS_WARNING) << "Connection refused; retrying in 2 seconds";
+      rtc::Thread::Current()->PostDelayedTask(
+          SafeTask(safety_.flag(), [this] { DoConnect(); }), kReconnectDelay);
+    } else {
+      Close();
+      callback_->OnDisconnected();
+    }
+  }
+}
diff --git a/third_party/libwebrtc/examples/peerconnection/client/peer_connection_client.h b/third_party/libwebrtc/examples/peerconnection/client/peer_connection_client.h
new file mode 100644
index 0000000000..8f9c5b6a75
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/client/peer_connection_client.h
@@ -0,0 +1,129 @@
+/*
+ *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_PEERCONNECTION_CLIENT_PEER_CONNECTION_CLIENT_H_
+#define EXAMPLES_PEERCONNECTION_CLIENT_PEER_CONNECTION_CLIENT_H_
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "rtc_base/net_helpers.h"
+#include "rtc_base/physical_socket_server.h"
+#include "rtc_base/third_party/sigslot/sigslot.h"
+
+typedef std::map<int, std::string> Peers;
+
+struct PeerConnectionClientObserver {
+  virtual void OnSignedIn() = 0;  // Called when we're logged on.
+  virtual void OnDisconnected() = 0;
+  virtual void OnPeerConnected(int id, const std::string& name) = 0;
+  virtual void OnPeerDisconnected(int peer_id) = 0;
+  virtual void OnMessageFromPeer(int peer_id, const std::string& message) = 0;
+  virtual void OnMessageSent(int err) = 0;
+  virtual void OnServerConnectionFailure() = 0;
+
+ protected:
+  virtual ~PeerConnectionClientObserver() {}
+};
+
+class PeerConnectionClient : public sigslot::has_slots<> {
+ public:
+  enum State {
+    NOT_CONNECTED,
+    RESOLVING,
+    SIGNING_IN,
+    CONNECTED,
+    SIGNING_OUT_WAITING,
+    SIGNING_OUT,
+  };
+
+  PeerConnectionClient();
+  ~PeerConnectionClient();
+
+  int id() const;
+  bool is_connected() const;
+  const Peers& peers() const;
+
+  void RegisterObserver(PeerConnectionClientObserver* callback);
+
+  void Connect(const std::string& server,
+               int port,
+               const std::string& client_name);
+
+  bool SendToPeer(int peer_id, const std::string& message);
+  bool SendHangUp(int peer_id);
+  bool IsSendingMessage();
+
+  bool SignOut();
+
+ protected:
+  void DoConnect();
+  void Close();
+  void InitSocketSignals();
+  bool ConnectControlSocket();
+  void OnConnect(rtc::Socket* socket);
+  void OnHangingGetConnect(rtc::Socket* socket);
+  void OnMessageFromPeer(int peer_id, const std::string& message);
+
+  // Quick and dirty support for parsing HTTP header values.
+  bool GetHeaderValue(const std::string& data,
+                      size_t eoh,
+                      const char* header_pattern,
+                      size_t* value);
+
+  bool GetHeaderValue(const std::string& data,
+                      size_t eoh,
+                      const char* header_pattern,
+                      std::string* value);
+
+  // Returns true if the whole response has been read.
+  bool ReadIntoBuffer(rtc::Socket* socket,
+                      std::string* data,
+                      size_t* content_length);
+
+  void OnRead(rtc::Socket* socket);
+
+  void OnHangingGetRead(rtc::Socket* socket);
+
+  // Parses a single line entry in the form "<name>,<id>,<connected>"
+  bool ParseEntry(const std::string& entry,
+                  std::string* name,
+                  int* id,
+                  bool* connected);
+
+  int GetResponseStatus(const std::string& response);
+
+  bool ParseServerResponse(const std::string& response,
+                           size_t content_length,
+                           size_t* peer_id,
+                           size_t* eoh);
+
+  void OnClose(rtc::Socket* socket, int err);
+
+  void OnResolveResult(rtc::AsyncResolverInterface* resolver);
+
+  PeerConnectionClientObserver* callback_;
+  rtc::SocketAddress server_address_;
+  rtc::AsyncResolver* resolver_;
+  std::unique_ptr<rtc::Socket> control_socket_;
+  std::unique_ptr<rtc::Socket> hanging_get_;
+  std::string onconnect_data_;
+  std::string control_data_;
+  std::string notification_data_;
+  std::string client_name_;
+  Peers peers_;
+  State state_;
+  int my_id_;
+  webrtc::ScopedTaskSafety safety_;
+};
+
+#endif  // EXAMPLES_PEERCONNECTION_CLIENT_PEER_CONNECTION_CLIENT_H_
diff --git a/third_party/libwebrtc/examples/peerconnection/server/data_socket.cc b/third_party/libwebrtc/examples/peerconnection/server/data_socket.cc
new file mode 100644
index 0000000000..855ebd8c0c
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/server/data_socket.cc
@@ -0,0 +1,299 @@
+/*
+ *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/peerconnection/server/data_socket.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#if defined(WEBRTC_POSIX)
+#include <unistd.h>
+#endif
+
+#include "examples/peerconnection/server/utils.h"
+#include "rtc_base/checks.h"
+
+static const char kHeaderTerminator[] = "\r\n\r\n";
+static const int kHeaderTerminatorLength = sizeof(kHeaderTerminator) - 1;
+
+// static
+const char DataSocket::kCrossOriginAllowHeaders[] =
+    "Access-Control-Allow-Origin: *\r\n"
+    "Access-Control-Allow-Credentials: true\r\n"
+    "Access-Control-Allow-Methods: POST, GET, OPTIONS\r\n"
+    "Access-Control-Allow-Headers: Content-Type, "
+    "Content-Length, Connection, Cache-Control\r\n"
+    "Access-Control-Expose-Headers: Content-Length\r\n";
+
+#if defined(WIN32)
+class WinsockInitializer {
+  static WinsockInitializer singleton;
+
+  WinsockInitializer() {
+    WSADATA data;
+    WSAStartup(MAKEWORD(1, 0), &data);
+  }
+
+ public:
+  ~WinsockInitializer() { WSACleanup(); }
+};
+WinsockInitializer WinsockInitializer::singleton;
+#endif
+
+//
+// SocketBase
+//
+
+bool SocketBase::Create() {
+  RTC_DCHECK(!valid());
+  socket_ = ::socket(AF_INET, SOCK_STREAM, 0);
+  return valid();
+}
+
+void SocketBase::Close() {
+  if (socket_ != INVALID_SOCKET) {
+    closesocket(socket_);
+    socket_ = INVALID_SOCKET;
+  }
+}
+
+//
+// DataSocket
+//
+
+std::string DataSocket::request_arguments() const {
+  size_t args = request_path_.find('?');
+  if (args != std::string::npos)
+    return request_path_.substr(args + 1);
+  return "";
+}
+
+bool DataSocket::PathEquals(const char* path) const {
+  RTC_DCHECK(path);
+  size_t args = request_path_.find('?');
+  if (args != std::string::npos)
+    return request_path_.substr(0, args).compare(path) == 0;
+  return request_path_.compare(path) == 0;
+}
+
+bool DataSocket::OnDataAvailable(bool* close_socket) {
+  RTC_DCHECK(valid());
+  char buffer[0xfff] = {0};
+  int bytes = recv(socket_, buffer, sizeof(buffer), 0);
+  if (bytes == SOCKET_ERROR || bytes == 0) {
+    *close_socket = true;
+    return false;
+  }
+
+  *close_socket = false;
+
+  bool ret = true;
+  if (headers_received()) {
+    if (method_ != POST) {
+      // unexpectedly received data.
+      ret = false;
+    } else {
+      data_.append(buffer, bytes);
+    }
+  } else {
+    request_headers_.append(buffer, bytes);
+    size_t found = request_headers_.find(kHeaderTerminator);
+    if (found != std::string::npos) {
+      data_ = request_headers_.substr(found + kHeaderTerminatorLength);
+      request_headers_.resize(found + kHeaderTerminatorLength);
+      ret = ParseHeaders();
+    }
+  }
+  return ret;
+}
+
+bool DataSocket::Send(const std::string& data) const {
+  return send(socket_, data.data(), static_cast<int>(data.length()), 0) !=
+         SOCKET_ERROR;
+}
+
+bool DataSocket::Send(const std::string& status,
+                      bool connection_close,
+                      const std::string& content_type,
+                      const std::string& extra_headers,
+                      const std::string& data) const {
+  RTC_DCHECK(valid());
+  RTC_DCHECK(!status.empty());
+  std::string buffer("HTTP/1.1 " + status + "\r\n");
+
+  buffer +=
+      "Server: PeerConnectionTestServer/0.1\r\n"
+      "Cache-Control: no-cache\r\n";
+
+  if (connection_close)
+    buffer += "Connection: close\r\n";
+
+  if (!content_type.empty())
+    buffer += "Content-Type: " + content_type + "\r\n";
+
+  buffer +=
+      "Content-Length: " + int2str(static_cast<int>(data.size())) + "\r\n";
+
+  if (!extra_headers.empty()) {
+    buffer += extra_headers;
+    // Extra headers are assumed to have a separator per header.
+  }
+
+  buffer += kCrossOriginAllowHeaders;
+
+  buffer += "\r\n";
+  buffer += data;
+
+  return Send(buffer);
+}
+
+void DataSocket::Clear() {
+  method_ = INVALID;
+  content_length_ = 0;
+  content_type_.clear();
+  request_path_.clear();
+  request_headers_.clear();
+  data_.clear();
+}
+
+bool DataSocket::ParseHeaders() {
+  RTC_DCHECK(!request_headers_.empty());
+  RTC_DCHECK_EQ(method_, INVALID);
+  size_t i = request_headers_.find("\r\n");
+  if (i == std::string::npos)
+    return false;
+
+  if (!ParseMethodAndPath(request_headers_.data(), i))
+    return false;
+
+  RTC_DCHECK_NE(method_, INVALID);
+  RTC_DCHECK(!request_path_.empty());
+
+  if (method_ == POST) {
+    const char* headers = request_headers_.data() + i + 2;
+    size_t len = request_headers_.length() - i - 2;
+    if (!ParseContentLengthAndType(headers, len))
+      return false;
+  }
+
+  return true;
+}
+
+bool DataSocket::ParseMethodAndPath(const char* begin, size_t len) {
+  struct {
+    const char* method_name;
+    size_t method_name_len;
+    RequestMethod id;
+  } supported_methods[] = {
+      {"GET", 3, GET},
+      {"POST", 4, POST},
+      {"OPTIONS", 7, OPTIONS},
+  };
+
+  const char* path = NULL;
+  for (size_t i = 0; i < ARRAYSIZE(supported_methods); ++i) {
+    if (len > supported_methods[i].method_name_len &&
+        isspace(begin[supported_methods[i].method_name_len]) &&
+        strncmp(begin, supported_methods[i].method_name,
+                supported_methods[i].method_name_len) == 0) {
+      method_ = supported_methods[i].id;
+      path = begin + supported_methods[i].method_name_len;
+      break;
+    }
+  }
+
+  const char* end = begin + len;
+  if (!path || path >= end)
+    return false;
+
+  ++path;
+  begin = path;
+  while (!isspace(*path) && path < end)
+    ++path;
+
+  request_path_.assign(begin, path - begin);
+
+  return true;
+}
+
+bool DataSocket::ParseContentLengthAndType(const char* headers, size_t length) {
+  RTC_DCHECK_EQ(content_length_, 0);
+  RTC_DCHECK(content_type_.empty());
+
+  const char* end = headers + length;
+  while (headers && headers < end) {
+    if (!isspace(headers[0])) {
+      static const char kContentLength[] = "Content-Length:";
+      static const char kContentType[] = "Content-Type:";
+      if ((headers + ARRAYSIZE(kContentLength)) < end &&
+          strncmp(headers, kContentLength, ARRAYSIZE(kContentLength) - 1) ==
+              0) {
+        headers += ARRAYSIZE(kContentLength) - 1;
+        while (headers[0] == ' ')
+          ++headers;
+        content_length_ = atoi(headers);
+      } else if ((headers + ARRAYSIZE(kContentType)) < end &&
+                 strncmp(headers, kContentType, ARRAYSIZE(kContentType) - 1) ==
+                     0) {
+        headers += ARRAYSIZE(kContentType) - 1;
+        while (headers[0] == ' ')
+          ++headers;
+        const char* type_end = strstr(headers, "\r\n");
+        if (type_end == NULL)
+          type_end = end;
+        content_type_.assign(headers, type_end);
+      }
+    } else {
+      ++headers;
+    }
+    headers = strstr(headers, "\r\n");
+    if (headers)
+      headers += 2;
+  }
+
+  return !content_type_.empty() && content_length_ != 0;
+}
+
+//
+// ListeningSocket
+//
+
+bool ListeningSocket::Listen(unsigned short port) {
+  RTC_DCHECK(valid());
+  int enabled = 1;
+  if (setsockopt(socket_, SOL_SOCKET, SO_REUSEADDR,
+                 reinterpret_cast<const char*>(&enabled),
+                 sizeof(enabled)) != 0) {
+    printf("setsockopt failed\n");
+    return false;
+  }
+  struct sockaddr_in addr = {0};
+  addr.sin_family = AF_INET;
+  addr.sin_addr.s_addr = htonl(INADDR_ANY);
+  addr.sin_port = htons(port);
+  if (bind(socket_, reinterpret_cast<const sockaddr*>(&addr), sizeof(addr)) ==
+      SOCKET_ERROR) {
+    printf("bind failed\n");
+    return false;
+  }
+  return listen(socket_, 5) != SOCKET_ERROR;
+}
+
+DataSocket* ListeningSocket::Accept() const {
+  RTC_DCHECK(valid());
+  struct sockaddr_in addr = {0};
+  socklen_t size = sizeof(addr);
+  NativeSocket client =
+      accept(socket_, reinterpret_cast<sockaddr*>(&addr), &size);
+  if (client == INVALID_SOCKET)
+    return NULL;
+
+  return new DataSocket(client);
+}
diff --git a/third_party/libwebrtc/examples/peerconnection/server/data_socket.h b/third_party/libwebrtc/examples/peerconnection/server/data_socket.h
new file mode 100644
index 0000000000..57ad5b9aee
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/server/data_socket.h
@@ -0,0 +1,152 @@
+/*
+ *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_PEERCONNECTION_SERVER_DATA_SOCKET_H_
+#define EXAMPLES_PEERCONNECTION_SERVER_DATA_SOCKET_H_
+
+#ifdef WIN32
+#include <winsock2.h>
+typedef int socklen_t;
+typedef SOCKET NativeSocket;
+#else
+#include <netinet/in.h>
+#include <sys/select.h>
+#include <sys/socket.h>
+#define closesocket close
+typedef int NativeSocket;
+
+#ifndef SOCKET_ERROR
+#define SOCKET_ERROR (-1)
+#endif
+
+#ifndef INVALID_SOCKET
+#define INVALID_SOCKET static_cast<NativeSocket>(-1)
+#endif
+#endif
+
+#include <string>
+
+class SocketBase {
+ public:
+  SocketBase() : socket_(INVALID_SOCKET) {}
+  explicit SocketBase(NativeSocket socket) : socket_(socket) {}
+  SocketBase(SocketBase& other) = delete;
+  SocketBase& operator=(const SocketBase& other) = delete;
+  ~SocketBase() { Close(); }
+
+  NativeSocket socket() const { return socket_; }
+  bool valid() const { return socket_ != INVALID_SOCKET; }
+
+  bool Create();
+  void Close();
+
+ protected:
+  NativeSocket socket_;
+};
+
+// Represents an HTTP server socket.
+class DataSocket : public SocketBase {
+ public:
+  enum RequestMethod {
+    INVALID,
+    GET,
+    POST,
+    OPTIONS,
+  };
+
+  explicit DataSocket(NativeSocket socket)
+      : SocketBase(socket), method_(INVALID), content_length_(0) {}
+
+  ~DataSocket() {}
+
+  static const char kCrossOriginAllowHeaders[];
+
+  bool headers_received() const { return method_ != INVALID; }
+
+  RequestMethod method() const { return method_; }
+
+  const std::string& request_path() const { return request_path_; }
+  std::string request_arguments() const;
+
+  const std::string& data() const { return data_; }
+
+  const std::string& content_type() const { return content_type_; }
+
+  size_t content_length() const { return content_length_; }
+
+  bool request_received() const {
+    return headers_received() && (method_ != POST || data_received());
+  }
+
+  bool data_received() const {
+    return method_ != POST || data_.length() >= content_length_;
+  }
+
+  // Checks if the request path (minus arguments) matches a given path.
+  bool PathEquals(const char* path) const;
+
+  // Called when we have received some data from clients.
+  // Returns false if an error occurred.
+  bool OnDataAvailable(bool* close_socket);
+
+  // Send a raw buffer of bytes.
+  bool Send(const std::string& data) const;
+
+  // Send an HTTP response.  The `status` should start with a valid HTTP
+  // response code, followed by a string.  E.g. "200 OK".
+  // If `connection_close` is set to true, an extra "Connection: close" HTTP
+  // header will be included.  `content_type` is the mime content type, not
+  // including the "Content-Type: " string.
+  // `extra_headers` should be either empty or a list of headers where each
+  // header terminates with "\r\n".
+  // `data` is the body of the message.  It's length will be specified via
+  // a "Content-Length" header.
+  bool Send(const std::string& status,
+            bool connection_close,
+            const std::string& content_type,
+            const std::string& extra_headers,
+            const std::string& data) const;
+
+  // Clears all held state and prepares the socket for receiving a new request.
+  void Clear();
+
+ protected:
+  // A fairly relaxed HTTP header parser.  Parses the method, path and
+  // content length (POST only) of a request.
+  // Returns true if a valid request was received and no errors occurred.
+  bool ParseHeaders();
+
+  // Figures out whether the request is a GET or POST and what path is
+  // being requested.
+  bool ParseMethodAndPath(const char* begin, size_t len);
+
+  // Determines the length of the body and it's mime type.
+  bool ParseContentLengthAndType(const char* headers, size_t length);
+
+ protected:
+  RequestMethod method_;
+  size_t content_length_;
+  std::string content_type_;
+  std::string request_path_;
+  std::string request_headers_;
+  std::string data_;
+};
+
+// The server socket.  Accepts connections and generates DataSocket instances
+// for each new connection.
+class ListeningSocket : public SocketBase {
+ public:
+  ListeningSocket() {}
+
+  bool Listen(unsigned short port);
+  DataSocket* Accept() const;
+};
+
+#endif  // EXAMPLES_PEERCONNECTION_SERVER_DATA_SOCKET_H_
diff --git a/third_party/libwebrtc/examples/peerconnection/server/main.cc b/third_party/libwebrtc/examples/peerconnection/server/main.cc
new file mode 100644
index 0000000000..50b8c23401
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/server/main.cc
@@ -0,0 +1,193 @@
+/*
+ *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#if defined(WEBRTC_POSIX)
+#include <sys/select.h>
+#endif
+#include <time.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "absl/flags/usage.h"
+#include "examples/peerconnection/server/data_socket.h"
+#include "examples/peerconnection/server/peer_channel.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/field_trial.h"
+
+ABSL_FLAG(
+    std::string,
+    force_fieldtrials,
+    "",
+    "Field trials control experimental features. This flag specifies the field "
+    "trials in effect. E.g. running with "
+    "--force_fieldtrials=WebRTC-FooFeature/Enabled/ "
+    "will assign the group Enabled to field trial WebRTC-FooFeature. Multiple "
+    "trials are separated by \"/\"");
+ABSL_FLAG(int, port, 8888, "default: 8888");
+
+static const size_t kMaxConnections = (FD_SETSIZE - 2);
+
+void HandleBrowserRequest(DataSocket* ds, bool* quit) {
+  RTC_DCHECK(ds && ds->valid());
+  RTC_DCHECK(quit);
+
+  const std::string& path = ds->request_path();
+
+  *quit = (path.compare("/quit") == 0);
+
+  if (*quit) {
+    ds->Send("200 OK", true, "text/html", "",
+             "<html><body>Quitting...</body></html>");
+  } else if (ds->method() == DataSocket::OPTIONS) {
+    // We'll get this when a browsers do cross-resource-sharing requests.
+    // The headers to allow cross-origin script support will be set inside
+    // Send.
+    ds->Send("200 OK", true, "", "", "");
+  } else {
+    // Here we could write some useful output back to the browser depending on
+    // the path.
+    printf("Received an invalid request: %s\n", ds->request_path().c_str());
+    ds->Send("500 Sorry", true, "text/html", "",
+             "<html><body>Sorry, not yet implemented</body></html>");
+  }
+}
+
+int main(int argc, char* argv[]) {
+  absl::SetProgramUsageMessage(
+      "Example usage: ./peerconnection_server --port=8888\n");
+  absl::ParseCommandLine(argc, argv);
+
+  // InitFieldTrialsFromString stores the char*, so the char array must outlive
+  // the application.
+  const std::string force_field_trials = absl::GetFlag(FLAGS_force_fieldtrials);
+  webrtc::field_trial::InitFieldTrialsFromString(force_field_trials.c_str());
+
+  int port = absl::GetFlag(FLAGS_port);
+
+  // Abort if the user specifies a port that is outside the allowed
+  // range [1, 65535].
+  if ((port < 1) || (port > 65535)) {
+    printf("Error: %i is not a valid port.\n", port);
+    return -1;
+  }
+
+  ListeningSocket listener;
+  if (!listener.Create()) {
+    printf("Failed to create server socket\n");
+    return -1;
+  } else if (!listener.Listen(port)) {
+    printf("Failed to listen on server socket\n");
+    return -1;
+  }
+
+  printf("Server listening on port %i\n", port);
+
+  PeerChannel clients;
+  typedef std::vector<DataSocket*> SocketArray;
+  SocketArray sockets;
+  bool quit = false;
+  while (!quit) {
+    fd_set socket_set;
+    FD_ZERO(&socket_set);
+    if (listener.valid())
+      FD_SET(listener.socket(), &socket_set);
+
+    for (SocketArray::iterator i = sockets.begin(); i != sockets.end(); ++i)
+      FD_SET((*i)->socket(), &socket_set);
+
+    struct timeval timeout = {10, 0};
+    if (select(FD_SETSIZE, &socket_set, NULL, NULL, &timeout) == SOCKET_ERROR) {
+      printf("select failed\n");
+      break;
+    }
+
+    for (SocketArray::iterator i = sockets.begin(); i != sockets.end(); ++i) {
+      DataSocket* s = *i;
+      bool socket_done = true;
+      if (FD_ISSET(s->socket(), &socket_set)) {
+        if (s->OnDataAvailable(&socket_done) && s->request_received()) {
+          ChannelMember* member = clients.Lookup(s);
+          if (member || PeerChannel::IsPeerConnection(s)) {
+            if (!member) {
+              if (s->PathEquals("/sign_in")) {
+                clients.AddMember(s);
+              } else {
+                printf("No member found for: %s\n", s->request_path().c_str());
+                s->Send("500 Error", true, "text/plain", "",
+                        "Peer most likely gone.");
+              }
+            } else if (member->is_wait_request(s)) {
+              // no need to do anything.
+              socket_done = false;
+            } else {
+              ChannelMember* target = clients.IsTargetedRequest(s);
+              if (target) {
+                member->ForwardRequestToPeer(s, target);
+              } else if (s->PathEquals("/sign_out")) {
+                s->Send("200 OK", true, "text/plain", "", "");
+              } else {
+                printf("Couldn't find target for request: %s\n",
+                       s->request_path().c_str());
+                s->Send("500 Error", true, "text/plain", "",
+                        "Peer most likely gone.");
+              }
+            }
+          } else {
+            HandleBrowserRequest(s, &quit);
+            if (quit) {
+              printf("Quitting...\n");
+              FD_CLR(listener.socket(), &socket_set);
+              listener.Close();
+              clients.CloseAll();
+            }
+          }
+        }
+      } else {
+        socket_done = false;
+      }
+
+      if (socket_done) {
+        printf("Disconnecting socket\n");
+        clients.OnClosing(s);
+        RTC_DCHECK(s->valid());  // Close must not have been called yet.
+        FD_CLR(s->socket(), &socket_set);
+        delete (*i);
+        i = sockets.erase(i);
+        if (i == sockets.end())
+          break;
+      }
+    }
+
+    clients.CheckForTimeout();
+
+    if (FD_ISSET(listener.socket(), &socket_set)) {
+      DataSocket* s = listener.Accept();
+      if (sockets.size() >= kMaxConnections) {
+        delete s;  // sorry, that's all we can take.
+        printf("Connection limit reached\n");
+      } else {
+        sockets.push_back(s);
+        printf("New connection...\n");
+      }
+    }
+  }
+
+  for (SocketArray::iterator i = sockets.begin(); i != sockets.end(); ++i)
+    delete (*i);
+  sockets.clear();
+
+  return 0;
+}
diff --git a/third_party/libwebrtc/examples/peerconnection/server/peer_channel.cc b/third_party/libwebrtc/examples/peerconnection/server/peer_channel.cc
new file mode 100644
index 0000000000..f53820cc60
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/server/peer_channel.cc
@@ -0,0 +1,360 @@
+/*
+ *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/peerconnection/server/peer_channel.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <algorithm>
+
+#include "examples/peerconnection/server/data_socket.h"
+#include "examples/peerconnection/server/utils.h"
+#include "rtc_base/checks.h"
+
+// Set to the peer id of the originator when messages are being
+// exchanged between peers, but set to the id of the receiving peer
+// itself when notifications are sent from the server about the state
+// of other peers.
+//
+// WORKAROUND: Since support for CORS varies greatly from one browser to the
+// next, we don't use a custom name for our peer-id header (originally it was
+// "X-Peer-Id: ").  Instead, we use a "simple header", "Pragma" which should
+// always be exposed to CORS requests.  There is a special CORS header devoted
+// to exposing proprietary headers (Access-Control-Expose-Headers), however
+// at this point it is not working correctly in some popular browsers.
+static const char kPeerIdHeader[] = "Pragma: ";
+
+static const char* kRequestPaths[] = {
+    "/wait",
+    "/sign_out",
+    "/message",
+};
+
+enum RequestPathIndex {
+  kWait,
+  kSignOut,
+  kMessage,
+};
+
+const size_t kMaxNameLength = 512;
+
+//
+// ChannelMember
+//
+
+int ChannelMember::s_member_id_ = 0;
+
+ChannelMember::ChannelMember(DataSocket* socket)
+    : waiting_socket_(NULL),
+      id_(++s_member_id_),
+      connected_(true),
+      timestamp_(time(NULL)) {
+  RTC_DCHECK(socket);
+  RTC_DCHECK_EQ(socket->method(), DataSocket::GET);
+  RTC_DCHECK(socket->PathEquals("/sign_in"));
+  name_ = socket->request_arguments();
+  if (name_.empty())
+    name_ = "peer_" + int2str(id_);
+  else if (name_.length() > kMaxNameLength)
+    name_.resize(kMaxNameLength);
+
+  std::replace(name_.begin(), name_.end(), ',', '_');
+}
+
+ChannelMember::~ChannelMember() {}
+
+bool ChannelMember::is_wait_request(DataSocket* ds) const {
+  return ds && ds->PathEquals(kRequestPaths[kWait]);
+}
+
+bool ChannelMember::TimedOut() {
+  return waiting_socket_ == NULL && (time(NULL) - timestamp_) > 30;
+}
+
+std::string ChannelMember::GetPeerIdHeader() const {
+  std::string ret(kPeerIdHeader + int2str(id_) + "\r\n");
+  return ret;
+}
+
+bool ChannelMember::NotifyOfOtherMember(const ChannelMember& other) {
+  RTC_DCHECK_NE(&other, this);
+  QueueResponse("200 OK", "text/plain", GetPeerIdHeader(), other.GetEntry());
+  return true;
+}
+
+// Returns a string in the form "name,id,connected\n".
+std::string ChannelMember::GetEntry() const {
+  RTC_DCHECK(name_.length() <= kMaxNameLength);
+
+  // name, 11-digit int, 1-digit bool, newline, null
+  char entry[kMaxNameLength + 15];
+  snprintf(entry, sizeof(entry), "%s,%d,%d\n",
+           name_.substr(0, kMaxNameLength).c_str(), id_, connected_);
+  return entry;
+}
+
+void ChannelMember::ForwardRequestToPeer(DataSocket* ds, ChannelMember* peer) {
+  RTC_DCHECK(peer);
+  RTC_DCHECK(ds);
+
+  std::string extra_headers(GetPeerIdHeader());
+
+  if (peer == this) {
+    ds->Send("200 OK", true, ds->content_type(), extra_headers, ds->data());
+  } else {
+    printf("Client %s sending to %s\n", name_.c_str(), peer->name().c_str());
+    peer->QueueResponse("200 OK", ds->content_type(), extra_headers,
+                        ds->data());
+    ds->Send("200 OK", true, "text/plain", "", "");
+  }
+}
+
+void ChannelMember::OnClosing(DataSocket* ds) {
+  if (ds == waiting_socket_) {
+    waiting_socket_ = NULL;
+    timestamp_ = time(NULL);
+  }
+}
+
+void ChannelMember::QueueResponse(const std::string& status,
+                                  const std::string& content_type,
+                                  const std::string& extra_headers,
+                                  const std::string& data) {
+  if (waiting_socket_) {
+    RTC_DCHECK(queue_.empty());
+    RTC_DCHECK_EQ(waiting_socket_->method(), DataSocket::GET);
+    bool ok =
+        waiting_socket_->Send(status, true, content_type, extra_headers, data);
+    if (!ok) {
+      printf("Failed to deliver data to waiting socket\n");
+    }
+    waiting_socket_ = NULL;
+    timestamp_ = time(NULL);
+  } else {
+    QueuedResponse qr;
+    qr.status = status;
+    qr.content_type = content_type;
+    qr.extra_headers = extra_headers;
+    qr.data = data;
+    queue_.push(qr);
+  }
+}
+
+void ChannelMember::SetWaitingSocket(DataSocket* ds) {
+  RTC_DCHECK_EQ(ds->method(), DataSocket::GET);
+  if (ds && !queue_.empty()) {
+    RTC_DCHECK(!waiting_socket_);
+    const QueuedResponse& response = queue_.front();
+    ds->Send(response.status, true, response.content_type,
+             response.extra_headers, response.data);
+    queue_.pop();
+  } else {
+    waiting_socket_ = ds;
+  }
+}
+
+//
+// PeerChannel
+//
+
+// static
+bool PeerChannel::IsPeerConnection(const DataSocket* ds) {
+  RTC_DCHECK(ds);
+  return (ds->method() == DataSocket::POST && ds->content_length() > 0) ||
+         (ds->method() == DataSocket::GET && ds->PathEquals("/sign_in"));
+}
+
+ChannelMember* PeerChannel::Lookup(DataSocket* ds) const {
+  RTC_DCHECK(ds);
+
+  if (ds->method() != DataSocket::GET && ds->method() != DataSocket::POST)
+    return NULL;
+
+  size_t i = 0;
+  for (; i < ARRAYSIZE(kRequestPaths); ++i) {
+    if (ds->PathEquals(kRequestPaths[i]))
+      break;
+  }
+
+  if (i == ARRAYSIZE(kRequestPaths))
+    return NULL;
+
+  std::string args(ds->request_arguments());
+  static const char kPeerId[] = "peer_id=";
+  size_t found = args.find(kPeerId);
+  if (found == std::string::npos)
+    return NULL;
+
+  int id = atoi(&args[found + ARRAYSIZE(kPeerId) - 1]);
+  Members::const_iterator iter = members_.begin();
+  for (; iter != members_.end(); ++iter) {
+    if (id == (*iter)->id()) {
+      if (i == kWait)
+        (*iter)->SetWaitingSocket(ds);
+      if (i == kSignOut)
+        (*iter)->set_disconnected();
+      return *iter;
+    }
+  }
+
+  return NULL;
+}
+
+ChannelMember* PeerChannel::IsTargetedRequest(const DataSocket* ds) const {
+  RTC_DCHECK(ds);
+  // Regardless of GET or POST, we look for the peer_id parameter
+  // only in the request_path.
+  const std::string& path = ds->request_path();
+  size_t args = path.find('?');
+  if (args == std::string::npos)
+    return NULL;
+  size_t found;
+  const char kTargetPeerIdParam[] = "to=";
+  do {
+    found = path.find(kTargetPeerIdParam, args);
+    if (found == std::string::npos)
+      return NULL;
+    if (found == (args + 1) || path[found - 1] == '&') {
+      found += ARRAYSIZE(kTargetPeerIdParam) - 1;
+      break;
+    }
+    args = found + ARRAYSIZE(kTargetPeerIdParam) - 1;
+  } while (true);
+  int id = atoi(&path[found]);
+  Members::const_iterator i = members_.begin();
+  for (; i != members_.end(); ++i) {
+    if ((*i)->id() == id) {
+      return *i;
+    }
+  }
+  return NULL;
+}
+
+bool PeerChannel::AddMember(DataSocket* ds) {
+  RTC_DCHECK(IsPeerConnection(ds));
+  ChannelMember* new_guy = new ChannelMember(ds);
+  Members failures;
+  BroadcastChangedState(*new_guy, &failures);
+  HandleDeliveryFailures(&failures);
+  members_.push_back(new_guy);
+
+  printf("New member added (total=%s): %s\n",
+         size_t2str(members_.size()).c_str(), new_guy->name().c_str());
+
+  // Let the newly connected peer know about other members of the channel.
+  std::string content_type;
+  std::string response = BuildResponseForNewMember(*new_guy, &content_type);
+  ds->Send("200 Added", true, content_type, new_guy->GetPeerIdHeader(),
+           response);
+  return true;
+}
+
+void PeerChannel::CloseAll() {
+  Members::const_iterator i = members_.begin();
+  for (; i != members_.end(); ++i) {
+    (*i)->QueueResponse("200 OK", "text/plain", "", "Server shutting down");
+  }
+  DeleteAll();
+}
+
+void PeerChannel::OnClosing(DataSocket* ds) {
+  for (Members::iterator i = members_.begin(); i != members_.end(); ++i) {
+    ChannelMember* m = (*i);
+    m->OnClosing(ds);
+    if (!m->connected()) {
+      i = members_.erase(i);
+      Members failures;
+      BroadcastChangedState(*m, &failures);
+      HandleDeliveryFailures(&failures);
+      delete m;
+      if (i == members_.end())
+        break;
+    }
+  }
+  printf("Total connected: %s\n", size_t2str(members_.size()).c_str());
+}
+
+void PeerChannel::CheckForTimeout() {
+  for (Members::iterator i = members_.begin(); i != members_.end(); ++i) {
+    ChannelMember* m = (*i);
+    if (m->TimedOut()) {
+      printf("Timeout: %s\n", m->name().c_str());
+      m->set_disconnected();
+      i = members_.erase(i);
+      Members failures;
+      BroadcastChangedState(*m, &failures);
+      HandleDeliveryFailures(&failures);
+      delete m;
+      if (i == members_.end())
+        break;
+    }
+  }
+}
+
+void PeerChannel::DeleteAll() {
+  for (Members::iterator i = members_.begin(); i != members_.end(); ++i)
+    delete (*i);
+  members_.clear();
+}
+
+void PeerChannel::BroadcastChangedState(const ChannelMember& member,
+                                        Members* delivery_failures) {
+  // This function should be called prior to DataSocket::Close().
+  RTC_DCHECK(delivery_failures);
+
+  if (!member.connected()) {
+    printf("Member disconnected: %s\n", member.name().c_str());
+  }
+
+  Members::iterator i = members_.begin();
+  for (; i != members_.end(); ++i) {
+    if (&member != (*i)) {
+      if (!(*i)->NotifyOfOtherMember(member)) {
+        (*i)->set_disconnected();
+        delivery_failures->push_back(*i);
+        i = members_.erase(i);
+        if (i == members_.end())
+          break;
+      }
+    }
+  }
+}
+
+void PeerChannel::HandleDeliveryFailures(Members* failures) {
+  RTC_DCHECK(failures);
+
+  while (!failures->empty()) {
+    Members::iterator i = failures->begin();
+    ChannelMember* member = *i;
+    RTC_DCHECK(!member->connected());
+    failures->erase(i);
+    BroadcastChangedState(*member, failures);
+    delete member;
+  }
+}
+
+// Builds a simple list of "name,id\n" entries for each member.
+std::string PeerChannel::BuildResponseForNewMember(const ChannelMember& member,
+                                                   std::string* content_type) {
+  RTC_DCHECK(content_type);
+
+  *content_type = "text/plain";
+  // The peer itself will always be the first entry.
+  std::string response(member.GetEntry());
+  for (Members::iterator i = members_.begin(); i != members_.end(); ++i) {
+    if (member.id() != (*i)->id()) {
+      RTC_DCHECK((*i)->connected());
+      response += (*i)->GetEntry();
+    }
+  }
+
+  return response;
+}
diff --git a/third_party/libwebrtc/examples/peerconnection/server/peer_channel.h b/third_party/libwebrtc/examples/peerconnection/server/peer_channel.h
new file mode 100644
index 0000000000..c3624908ac
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/server/peer_channel.h
@@ -0,0 +1,118 @@
+/*
+ *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_PEERCONNECTION_SERVER_PEER_CHANNEL_H_
+#define EXAMPLES_PEERCONNECTION_SERVER_PEER_CHANNEL_H_
+
+#include <time.h>
+
+#include <queue>
+#include <string>
+#include <vector>
+
+class DataSocket;
+
+// Represents a single peer connected to the server.
+class ChannelMember {
+ public:
+  explicit ChannelMember(DataSocket* socket);
+  ~ChannelMember();
+
+  bool connected() const { return connected_; }
+  int id() const { return id_; }
+  void set_disconnected() { connected_ = false; }
+  bool is_wait_request(DataSocket* ds) const;
+  const std::string& name() const { return name_; }
+
+  bool TimedOut();
+
+  std::string GetPeerIdHeader() const;
+
+  bool NotifyOfOtherMember(const ChannelMember& other);
+
+  // Returns a string in the form "name,id\n".
+  std::string GetEntry() const;
+
+  void ForwardRequestToPeer(DataSocket* ds, ChannelMember* peer);
+
+  void OnClosing(DataSocket* ds);
+
+  void QueueResponse(const std::string& status,
+                     const std::string& content_type,
+                     const std::string& extra_headers,
+                     const std::string& data);
+
+  void SetWaitingSocket(DataSocket* ds);
+
+ protected:
+  struct QueuedResponse {
+    std::string status, content_type, extra_headers, data;
+  };
+
+  DataSocket* waiting_socket_;
+  int id_;
+  bool connected_;
+  time_t timestamp_;
+  std::string name_;
+  std::queue<QueuedResponse> queue_;
+  static int s_member_id_;
+};
+
+// Manages all currently connected peers.
+class PeerChannel {
+ public:
+  typedef std::vector<ChannelMember*> Members;
+
+  PeerChannel() {}
+
+  ~PeerChannel() { DeleteAll(); }
+
+  const Members& members() const { return members_; }
+
+  // Returns true if the request should be treated as a new ChannelMember
+  // request.  Otherwise the request is not peerconnection related.
+  static bool IsPeerConnection(const DataSocket* ds);
+
+  // Finds a connected peer that's associated with the `ds` socket.
+  ChannelMember* Lookup(DataSocket* ds) const;
+
+  // Checks if the request has a "peer_id" parameter and if so, looks up the
+  // peer for which the request is targeted at.
+  ChannelMember* IsTargetedRequest(const DataSocket* ds) const;
+
+  // Adds a new ChannelMember instance to the list of connected peers and
+  // associates it with the socket.
+  bool AddMember(DataSocket* ds);
+
+  // Closes all connections and sends a "shutting down" message to all
+  // connected peers.
+  void CloseAll();
+
+  // Called when a socket was determined to be closing by the peer (or if the
+  // connection went dead).
+  void OnClosing(DataSocket* ds);
+
+  void CheckForTimeout();
+
+ protected:
+  void DeleteAll();
+  void BroadcastChangedState(const ChannelMember& member,
+                             Members* delivery_failures);
+  void HandleDeliveryFailures(Members* failures);
+
+  // Builds a simple list of "name,id\n" entries for each member.
+  std::string BuildResponseForNewMember(const ChannelMember& member,
+                                        std::string* content_type);
+
+ protected:
+  Members members_;
+};
+
+#endif  // EXAMPLES_PEERCONNECTION_SERVER_PEER_CHANNEL_H_
diff --git a/third_party/libwebrtc/examples/peerconnection/server/server_test.html b/third_party/libwebrtc/examples/peerconnection/server/server_test.html
new file mode 100644
index 0000000000..0a165f19d5
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/server/server_test.html
@@ -0,0 +1,237 @@
+<html>
+<head>
+<title>PeerConnection server test page</title>
+
+<script>
+var request = null;
+var hangingGet = null;
+var localName;
+var server;
+var my_id = -1;
+var other_peers = {};
+var message_counter = 0;
+
+function trace(txt) {
+  var elem = document.getElementById("debug");
+  elem.innerHTML += txt + "<br>";
+}
+
+function handleServerNotification(data) {
+  trace("Server notification: " + data);
+  var parsed = data.split(',');
+  if (parseInt(parsed[2]) != 0)
+    other_peers[parseInt(parsed[1])] = parsed[0];
+}
+
+function handlePeerMessage(peer_id, data) {
+  ++message_counter;
+  var str = "Message from '" + other_peers[peer_id] + "'&nbsp;";
+  str += "<span id='toggle_" + message_counter + "' onclick='toggleMe(this);' ";
+  str += "style='cursor: pointer'>+</span><br>";
+  str += "<blockquote id='msg_" + message_counter + "' style='display:none'>";
+  str += data + "</blockquote>";
+  trace(str);
+  if (document.getElementById("loopback").checked) {
+    if (data.search("offer") != -1) {
+      // In loopback mode, if DTLS is enabled, notify the client to disable it.
+      // Otherwise replace the offer with an answer.
+      if (data.search("fingerprint") != -1)
+        data = data.replace("offer", "offer-loopback");
+      else
+        data = data.replace("offer", "answer");
+    }
+    sendToPeer(peer_id, data);
+  }
+}
+
+function GetIntHeader(r, name) {
+  var val = r.getResponseHeader(name);
+  return val != null && val.length ? parseInt(val) : -1;
+}
+
+function hangingGetCallback() {
+  try {
+    if (hangingGet.readyState != 4)
+      return;
+    if (hangingGet.status != 200) {
+      trace("server error: " + hangingGet.statusText);
+      disconnect();
+    } else {
+      var peer_id = GetIntHeader(hangingGet, "Pragma");
+      if (peer_id == my_id) {
+        handleServerNotification(hangingGet.responseText);
+      } else {
+        handlePeerMessage(peer_id, hangingGet.responseText);
+      }
+    }
+
+    if (hangingGet) {
+      hangingGet.abort();
+      hangingGet = null;
+    }
+
+    if (my_id != -1)
+      window.setTimeout(startHangingGet, 0);
+  } catch (e) {
+    trace("Hanging get error: " + e.description);
+  }
+}
+
+function startHangingGet() {
+  try {
+    hangingGet = new XMLHttpRequest();
+    hangingGet.onreadystatechange = hangingGetCallback;
+    hangingGet.ontimeout = onHangingGetTimeout;
+    hangingGet.open("GET", server + "/wait?peer_id=" + my_id, true);
+    hangingGet.send();  
+  } catch (e) {
+    trace("error" + e.description);
+  }
+}
+
+function onHangingGetTimeout() {
+  trace("hanging get timeout. issuing again.");
+  hangingGet.abort();
+  hangingGet = null;
+  if (my_id != -1)
+    window.setTimeout(startHangingGet, 0);
+}
+
+function signInCallback() {
+  try {
+    if (request.readyState == 4) {
+      if (request.status == 200) {
+        var peers = request.responseText.split("\n");
+        my_id = parseInt(peers[0].split(',')[1]);
+        trace("My id: " + my_id);
+        for (var i = 1; i < peers.length; ++i) {
+          if (peers[i].length > 0) {
+            trace("Peer " + i + ": " + peers[i]);
+            var parsed = peers[i].split(',');
+            other_peers[parseInt(parsed[1])] = parsed[0];
+          }
+        }
+        startHangingGet();
+        request = null;
+      }
+    }
+  } catch (e) {
+    trace("error: " + e.description);
+  }
+}
+
+function signIn() {
+  try {
+    request = new XMLHttpRequest();
+    request.onreadystatechange = signInCallback;
+    request.open("GET", server + "/sign_in?" + localName, true);
+    request.send();
+  } catch (e) {
+    trace("error: " + e.description);
+  }
+}
+
+function sendToPeer(peer_id, data) {
+  if (my_id == -1) {
+    alert("Not connected");
+    return;
+  }
+  if (peer_id == my_id) {
+    alert("Can't send a message to oneself :)");
+    return;
+  }
+  var r = new XMLHttpRequest();
+  r.open("POST", server + "/message?peer_id=" + my_id + "&to=" + peer_id,
+         false);
+  r.setRequestHeader("Content-Type", "text/plain");
+  r.send(data);
+  r = null;
+}
+
+function connect() {
+  localName = document.getElementById("local").value.toLowerCase();
+  server = document.getElementById("server").value.toLowerCase();
+  if (localName.length == 0) {
+    alert("I need a name please.");
+    document.getElementById("local").focus();
+  } else {
+    document.getElementById("connect").disabled = true;
+    document.getElementById("disconnect").disabled = false;
+    document.getElementById("send").disabled = false;
+    signIn();
+  }
+}
+
+function disconnect() {
+  if (request) {
+    request.abort();
+    request = null;
+  }
+  
+  if (hangingGet) {
+    hangingGet.abort();
+    hangingGet = null;
+  }
+
+  if (my_id != -1) {
+    request = new XMLHttpRequest();
+    request.open("GET", server + "/sign_out?peer_id=" + my_id, false);
+    request.send();
+    request = null;
+    my_id = -1;
+  }
+
+  document.getElementById("connect").disabled = false;
+  document.getElementById("disconnect").disabled = true;
+  document.getElementById("send").disabled = true;
+}
+
+window.onbeforeunload = disconnect;
+
+function send() {
+  var text = document.getElementById("message").value;
+  var peer_id = parseInt(document.getElementById("peer_id").value);
+  if (!text.length || peer_id == 0) {
+    alert("No text supplied or invalid peer id");
+  } else {
+    sendToPeer(peer_id, text);
+  }
+}
+
+function toggleMe(obj) {
+  var id = obj.id.replace("toggle", "msg");
+  var t = document.getElementById(id);
+  if (obj.innerText == "+") {
+    obj.innerText = "-";
+    t.style.display = "block";
+  } else {
+    obj.innerText = "+";
+    t.style.display = "none";
+  }
+}
+
+</script>
+
+</head>
+<body>
+Server: <input type="text" id="server" value="http://localhost:8888" /><br>
+<input type="checkbox" id="loopback" checked="checked"/> Loopback (just send
+received messages right back)<br>
+Your name: <input type="text" id="local" value="my_name"/>
+<button id="connect" onclick="connect();">Connect</button>
+<button disabled="true" id="disconnect"
+        onclick="disconnect();">Disconnect</button>
+<br>
+<table><tr><td>
+Target peer id: <input type="text" id="peer_id" size="3"/></td><td>
+Message: <input type="text" id="message"/></td><td>
+<button disabled="true" id="send" onclick="send();">Send</button>
+</td></tr></table>
+<button onclick="document.getElementById('debug').innerHTML='';">
+Clear log</button>
+
+<pre id="debug">
+</pre>
+<br><hr>
+</body>
+</html>
diff --git a/third_party/libwebrtc/examples/peerconnection/server/utils.cc b/third_party/libwebrtc/examples/peerconnection/server/utils.cc
new file mode 100644
index 0000000000..5e61e601d9
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/server/utils.cc
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/peerconnection/server/utils.h"
+
+#include <stdio.h>
+
+#include "rtc_base/string_encode.h"
+
+using rtc::ToString;
+
+std::string int2str(int i) {
+  return ToString(i);
+}
+
+std::string size_t2str(size_t i) {
+  return ToString(i);
+}
diff --git a/third_party/libwebrtc/examples/peerconnection/server/utils.h b/third_party/libwebrtc/examples/peerconnection/server/utils.h
new file mode 100644
index 0000000000..85c04a40e9
--- /dev/null
+++ b/third_party/libwebrtc/examples/peerconnection/server/utils.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_PEERCONNECTION_SERVER_UTILS_H_
+#define EXAMPLES_PEERCONNECTION_SERVER_UTILS_H_
+
+#include <stddef.h>
+
+#include <string>
+
+#ifndef ARRAYSIZE
+#define ARRAYSIZE(x) (sizeof(x) / sizeof(x[0]))
+#endif
+
+std::string int2str(int i);
+std::string size_t2str(size_t i);
+
+#endif  // EXAMPLES_PEERCONNECTION_SERVER_UTILS_H_
diff --git a/third_party/libwebrtc/examples/stunprober/main.cc b/third_party/libwebrtc/examples/stunprober/main.cc
new file mode 100644
index 0000000000..3b3c06be8f
--- /dev/null
+++ b/third_party/libwebrtc/examples/stunprober/main.cc
@@ -0,0 +1,146 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <set>
+#include <sstream>
+#include <string>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "p2p/base/basic_packet_socket_factory.h"
+#include "p2p/stunprober/stun_prober.h"
+#include "rtc_base/helpers.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/network.h"
+#include "rtc_base/physical_socket_server.h"
+#include "rtc_base/socket_address.h"
+#include "rtc_base/ssl_adapter.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/time_utils.h"
+#include "test/scoped_key_value_config.h"
+
+using stunprober::AsyncCallback;
+using stunprober::StunProber;
+
+ABSL_FLAG(int,
+          interval,
+          10,
+          "Interval of consecutive stun pings in milliseconds");
+ABSL_FLAG(bool,
+          shared_socket,
+          false,
+          "Share socket mode for different remote IPs");
+ABSL_FLAG(int,
+          pings_per_ip,
+          10,
+          "Number of consecutive stun pings to send for each IP");
+ABSL_FLAG(int,
+          timeout,
+          1000,
+          "Milliseconds of wait after the last ping sent before exiting");
+ABSL_FLAG(
+    std::string,
+    servers,
+    "stun.l.google.com:19302,stun1.l.google.com:19302,stun2.l.google.com:19302",
+    "Comma separated STUN server addresses with ports");
+
+namespace {
+
+const char* PrintNatType(stunprober::NatType type) {
+  switch (type) {
+    case stunprober::NATTYPE_NONE:
+      return "Not behind a NAT";
+    case stunprober::NATTYPE_UNKNOWN:
+      return "Unknown NAT type";
+    case stunprober::NATTYPE_SYMMETRIC:
+      return "Symmetric NAT";
+    case stunprober::NATTYPE_NON_SYMMETRIC:
+      return "Non-Symmetric NAT";
+    default:
+      return "Invalid";
+  }
+}
+
+void PrintStats(StunProber* prober) {
+  StunProber::Stats stats;
+  if (!prober->GetStats(&stats)) {
+    RTC_LOG(LS_WARNING) << "Results are inconclusive.";
+    return;
+  }
+
+  RTC_LOG(LS_INFO) << "Shared Socket Mode: " << stats.shared_socket_mode;
+  RTC_LOG(LS_INFO) << "Requests sent: " << stats.num_request_sent;
+  RTC_LOG(LS_INFO) << "Responses received: " << stats.num_response_received;
+  RTC_LOG(LS_INFO) << "Target interval (ns): "
+                   << stats.target_request_interval_ns;
+  RTC_LOG(LS_INFO) << "Actual interval (ns): "
+                   << stats.actual_request_interval_ns;
+  RTC_LOG(LS_INFO) << "NAT Type: " << PrintNatType(stats.nat_type);
+  RTC_LOG(LS_INFO) << "Host IP: " << stats.host_ip;
+  RTC_LOG(LS_INFO) << "Server-reflexive ips: ";
+  for (auto& ip : stats.srflx_addrs) {
+    RTC_LOG(LS_INFO) << "\t" << ip;
+  }
+
+  RTC_LOG(LS_INFO) << "Success Precent: " << stats.success_percent;
+  RTC_LOG(LS_INFO) << "Response Latency:" << stats.average_rtt_ms;
+}
+
+void StopTrial(rtc::Thread* thread, StunProber* prober, int result) {
+  thread->Quit();
+  if (prober) {
+    RTC_LOG(LS_INFO) << "Result: " << result;
+    if (result == StunProber::SUCCESS) {
+      PrintStats(prober);
+    }
+  }
+}
+
+}  // namespace
+
+int main(int argc, char* argv[]) {
+  absl::ParseCommandLine(argc, argv);
+
+  std::vector<rtc::SocketAddress> server_addresses;
+  std::istringstream servers(absl::GetFlag(FLAGS_servers));
+  std::string server;
+  while (getline(servers, server, ',')) {
+    rtc::SocketAddress addr;
+    if (!addr.FromString(server)) {
+      RTC_LOG(LS_ERROR) << "Parsing " << server << " failed.";
+      return -1;
+    }
+    server_addresses.push_back(addr);
+  }
+
+  rtc::InitializeSSL();
+  rtc::InitRandom(rtc::Time32());
+  webrtc::test::ScopedKeyValueConfig field_trials;
+  rtc::PhysicalSocketServer socket_server;
+  rtc::AutoSocketServerThread thread(&socket_server);
+  auto socket_factory =
+      std::make_unique<rtc::BasicPacketSocketFactory>(&socket_server);
+  std::unique_ptr<rtc::BasicNetworkManager> network_manager(
+      new rtc::BasicNetworkManager(&socket_server, &field_trials));
+  std::vector<const rtc::Network*> networks = network_manager->GetNetworks();
+  auto prober = std::make_unique<StunProber>(socket_factory.get(),
+                                             rtc::Thread::Current(), networks);
+  auto finish_callback = [&thread](StunProber* prober, int result) {
+    StopTrial(&thread, prober, result);
+  };
+  prober->Start(server_addresses, absl::GetFlag(FLAGS_shared_socket),
+                absl::GetFlag(FLAGS_interval),
+                absl::GetFlag(FLAGS_pings_per_ip), absl::GetFlag(FLAGS_timeout),
+                AsyncCallback(finish_callback));
+  thread.Run();
+  return 0;
+}
diff --git a/third_party/libwebrtc/examples/stunserver/stunserver_main.cc b/third_party/libwebrtc/examples/stunserver/stunserver_main.cc
new file mode 100644
index 0000000000..8180069bf0
--- /dev/null
+++ b/third_party/libwebrtc/examples/stunserver/stunserver_main.cc
@@ -0,0 +1,49 @@
+/*
+ *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <iostream>
+
+#include "p2p/base/stun_server.h"
+#include "rtc_base/async_udp_socket.h"
+#include "rtc_base/socket_address.h"
+#include "rtc_base/socket_server.h"
+#include "rtc_base/thread.h"
+
+using cricket::StunServer;
+
+int main(int argc, char* argv[]) {
+  if (argc != 2) {
+    std::cerr << "usage: stunserver address" << std::endl;
+    return 1;
+  }
+
+  rtc::SocketAddress server_addr;
+  if (!server_addr.FromString(argv[1])) {
+    std::cerr << "Unable to parse IP address: " << argv[1];
+    return 1;
+  }
+
+  rtc::Thread* pthMain = rtc::Thread::Current();
+
+  rtc::AsyncUDPSocket* server_socket =
+      rtc::AsyncUDPSocket::Create(pthMain->socketserver(), server_addr);
+  if (!server_socket) {
+    std::cerr << "Failed to create a UDP socket" << std::endl;
+    return 1;
+  }
+
+  StunServer* server = new StunServer(server_socket);
+
+  std::cout << "Listening at " << server_addr.ToString() << std::endl;
+
+  pthMain->Run();
+
+  delete server;
+  return 0;
+}
diff --git a/third_party/libwebrtc/examples/turnserver/read_auth_file.cc b/third_party/libwebrtc/examples/turnserver/read_auth_file.cc
new file mode 100644
index 0000000000..4b0b21b8ae
--- /dev/null
+++ b/third_party/libwebrtc/examples/turnserver/read_auth_file.cc
@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/turnserver/read_auth_file.h"
+
+#include <stddef.h>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "rtc_base/string_encode.h"
+
+namespace webrtc_examples {
+
+std::map<std::string, std::string> ReadAuthFile(std::istream* s) {
+  std::map<std::string, std::string> name_to_key;
+  for (std::string line; std::getline(*s, line);) {
+    const size_t sep = line.find('=');
+    if (sep == std::string::npos)
+      continue;
+    char buf[32];
+    size_t len = rtc::hex_decode(rtc::ArrayView<char>(buf),
+                                 absl::string_view(line).substr(sep + 1));
+    if (len > 0) {
+      name_to_key.emplace(line.substr(0, sep), std::string(buf, len));
+    }
+  }
+  return name_to_key;
+}
+
+}  // namespace webrtc_examples
diff --git a/third_party/libwebrtc/examples/turnserver/read_auth_file.h b/third_party/libwebrtc/examples/turnserver/read_auth_file.h
new file mode 100644
index 0000000000..1c139c9924
--- /dev/null
+++ b/third_party/libwebrtc/examples/turnserver/read_auth_file.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_TURNSERVER_READ_AUTH_FILE_H_
+#define EXAMPLES_TURNSERVER_READ_AUTH_FILE_H_
+
+#include <istream>
+#include <map>
+#include <string>
+
+namespace webrtc_examples {
+
+std::map<std::string, std::string> ReadAuthFile(std::istream* s);
+
+}  // namespace webrtc_examples
+
+#endif  // EXAMPLES_TURNSERVER_READ_AUTH_FILE_H_
diff --git a/third_party/libwebrtc/examples/turnserver/read_auth_file_unittest.cc b/third_party/libwebrtc/examples/turnserver/read_auth_file_unittest.cc
new file mode 100644
index 0000000000..23b026429b
--- /dev/null
+++ b/third_party/libwebrtc/examples/turnserver/read_auth_file_unittest.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/turnserver/read_auth_file.h"
+
+#include <sstream>
+
+#include "test/gtest.h"
+
+namespace webrtc_examples {
+
+TEST(ReadAuthFile, HandlesEmptyFile) {
+  std::istringstream empty;
+  auto map = ReadAuthFile(&empty);
+  EXPECT_TRUE(map.empty());
+}
+
+TEST(ReadAuthFile, RecognizesValidUser) {
+  std::istringstream file("foo=deadbeaf\n");
+  auto map = ReadAuthFile(&file);
+  ASSERT_NE(map.find("foo"), map.end());
+  EXPECT_EQ(map["foo"], "\xde\xad\xbe\xaf");
+}
+
+TEST(ReadAuthFile, EmptyValueForInvalidHex) {
+  std::istringstream file(
+      "foo=deadbeaf\n"
+      "bar=xxxxinvalidhex\n"
+      "baz=cafe\n");
+  auto map = ReadAuthFile(&file);
+  ASSERT_NE(map.find("foo"), map.end());
+  EXPECT_EQ(map["foo"], "\xde\xad\xbe\xaf");
+  EXPECT_EQ(map.find("bar"), map.end());
+  ASSERT_NE(map.find("baz"), map.end());
+  EXPECT_EQ(map["baz"], "\xca\xfe");
+}
+
+}  // namespace webrtc_examples
diff --git a/third_party/libwebrtc/examples/turnserver/turnserver_main.cc b/third_party/libwebrtc/examples/turnserver/turnserver_main.cc
new file mode 100644
index 0000000000..8db6162306
--- /dev/null
+++ b/third_party/libwebrtc/examples/turnserver/turnserver_main.cc
@@ -0,0 +1,101 @@
+/*
+ *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <fstream>
+#include <iostream>
+#include <map>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "examples/turnserver/read_auth_file.h"
+#include "p2p/base/basic_packet_socket_factory.h"
+#include "p2p/base/port_interface.h"
+#include "p2p/base/turn_server.h"
+#include "rtc_base/async_udp_socket.h"
+#include "rtc_base/ip_address.h"
+#include "rtc_base/physical_socket_server.h"
+#include "rtc_base/socket_address.h"
+#include "rtc_base/thread.h"
+
+namespace {
+const char kSoftware[] = "libjingle TurnServer";
+
+class TurnFileAuth : public cricket::TurnAuthInterface {
+ public:
+  explicit TurnFileAuth(std::map<std::string, std::string> name_to_key)
+      : name_to_key_(std::move(name_to_key)) {}
+
+  virtual bool GetKey(absl::string_view username,
+                      absl::string_view realm,
+                      std::string* key) {
+    // File is stored as lines of <username>=<HA1>.
+    // Generate HA1 via "echo -n "<username>:<realm>:<password>" | md5sum"
+    auto it = name_to_key_.find(std::string(username));
+    if (it == name_to_key_.end())
+      return false;
+    *key = it->second;
+    return true;
+  }
+
+ private:
+  const std::map<std::string, std::string> name_to_key_;
+};
+
+}  // namespace
+
+int main(int argc, char* argv[]) {
+  if (argc != 5) {
+    std::cerr << "usage: turnserver int-addr ext-ip realm auth-file"
+              << std::endl;
+    return 1;
+  }
+
+  rtc::SocketAddress int_addr;
+  if (!int_addr.FromString(argv[1])) {
+    std::cerr << "Unable to parse IP address: " << argv[1] << std::endl;
+    return 1;
+  }
+
+  rtc::IPAddress ext_addr;
+  if (!IPFromString(argv[2], &ext_addr)) {
+    std::cerr << "Unable to parse IP address: " << argv[2] << std::endl;
+    return 1;
+  }
+
+  rtc::PhysicalSocketServer socket_server;
+  rtc::AutoSocketServerThread main(&socket_server);
+  rtc::AsyncUDPSocket* int_socket =
+      rtc::AsyncUDPSocket::Create(&socket_server, int_addr);
+  if (!int_socket) {
+    std::cerr << "Failed to create a UDP socket bound at" << int_addr.ToString()
+              << std::endl;
+    return 1;
+  }
+
+  cricket::TurnServer server(&main);
+  std::fstream auth_file(argv[4], std::fstream::in);
+
+  TurnFileAuth auth(auth_file.is_open()
+                        ? webrtc_examples::ReadAuthFile(&auth_file)
+                        : std::map<std::string, std::string>());
+  server.set_realm(argv[3]);
+  server.set_software(kSoftware);
+  server.set_auth_hook(&auth);
+  server.AddInternalSocket(int_socket, cricket::PROTO_UDP);
+  server.SetExternalSocketFactory(
+      new rtc::BasicPacketSocketFactory(&socket_server),
+      rtc::SocketAddress(ext_addr, 0));
+
+  std::cout << "Listening internally at " << int_addr.ToString() << std::endl;
+
+  main.Run();
+  return 0;
+}
diff --git a/third_party/libwebrtc/examples/unityplugin/ANDROID_INSTRUCTION b/third_party/libwebrtc/examples/unityplugin/ANDROID_INSTRUCTION
new file mode 100644
index 0000000000..d5f7399bca
--- /dev/null
+++ b/third_party/libwebrtc/examples/unityplugin/ANDROID_INSTRUCTION
@@ -0,0 +1,33 @@
+Instruction of running webrtc_unity_plugin on Android Unity
+
+1. On Linux machine, compile target webrtc_unity_plugin.
+   Checkout WebRTC codebase: fetch --nohooks webrtc_android
+   If you already have a checkout for linux, add target_os=”android” into .gclient file.
+      Run gclient sync
+      Run gn args out/Android, and again set target_os=”android” in the args.gn
+      Run ninja -C out/Android webrtc_unity_plugin
+
+2. On Linux machine, build target libwebrtc_unity under webrtc checkout. This is the java code for webrtc to work on Android.
+
+3. Copy libwebrtc_unity.jar and libwebrtc_unity_plugin.so into Unity project folder, under Assets/Plugins/Android folder.
+
+4. Rename libwebrtc_unity_plugin.so to libjingle_peerconnection_so.so. This is hacky, and the purpose is to let the java code in libwebrtc_unity.jar to find their JNI implementations. Simultaneously, in your C# wrapper script for the native plugin libjingle_peerconnection_so.so, the dll_path should be set to “jingle_peerconnection_so”.
+
+5. In the Unity Main Scene’s Start method, write the following code to initialize the Java environment for webrtc (otherwise, webrtc will not be able to access audio device or camera from C++ code):
+
+#if UNITY_ANDROID
+    AndroidJavaClass playerClass = new AndroidJavaClass("com.unity3d.player.UnityPlayer");
+    AndroidJavaObject activity = playerClass.GetStatic<AndroidJavaObject>("currentActivity");
+    AndroidJavaClass utilityClass = new AndroidJavaClass("org.webrtc.UnityUtility");
+    utilityClass.CallStatic("InitializePeerConncectionFactory", new object[1] { activity });
+#endif
+
+6. Compile the unity project into an APK, and decompile the apk using apktool that you can download from https://ibotpeaches.github.io/Apktool/
+    Run apktool d apkname.apk.
+Then copy the AndroidManifest.xml in the decompiled folder to the Assets/Plugins/Android folder, and add two lines:
+  <uses-permission android:name="android.permission.RECORD_AUDIO" />
+  <uses-permission android:name="android.permission.CAMERA" />
+
+The purpose of using apktool is to get a well-written android manifest xml file. If you know how to write manifest file from scratch, you can skip using apktool.
+
+7. Compile the unity project into an APK again and deploy it to an android device.
diff --git a/third_party/libwebrtc/examples/unityplugin/DEPS b/third_party/libwebrtc/examples/unityplugin/DEPS
new file mode 100644
index 0000000000..604005ac73
--- /dev/null
+++ b/third_party/libwebrtc/examples/unityplugin/DEPS
@@ -0,0 +1,4 @@
+include_rules = [
+  "+modules/utility",
+  "+sdk",
+]
diff --git a/third_party/libwebrtc/examples/unityplugin/README b/third_party/libwebrtc/examples/unityplugin/README
new file mode 100644
index 0000000000..da8f07aa11
--- /dev/null
+++ b/third_party/libwebrtc/examples/unityplugin/README
@@ -0,0 +1,309 @@
+This directory contains an example Unity native plugin for Windows OS and Android.
+
+The APIs use Platform Invoke (P/Invoke) technology as required by Unity native plugin.
+This plugin dll can also be used by Windows C# applications other than Unity.
+
+For detailed build instruction on Android, see ANDROID_INSTRUCTION
+
+An example of wrapping native plugin into a C# managed class in Unity is given as following:
+
+using System;
+using System.Collections.Generic;
+using System.Runtime.InteropServices;
+
+namespace SimplePeerConnectionM {
+  // A class for ice candidate.
+  public class IceCandidate {
+    public IceCandidate(string candidate, int sdpMlineIndex, string sdpMid) {
+      mCandidate = candidate;
+      mSdpMlineIndex = sdpMlineIndex;
+      mSdpMid = sdpMid;
+    }
+    string mCandidate;
+    int mSdpMlineIndex;
+    string mSdpMid;
+
+    public string Candidate {
+      get { return mCandidate; }
+      set { mCandidate = value; }
+    }
+
+    public int SdpMlineIndex {
+      get { return mSdpMlineIndex; }
+      set { mSdpMlineIndex = value; }
+    }
+
+    public string SdpMid {
+      get { return mSdpMid; }
+      set { mSdpMid = value; }
+    }
+  }
+
+  // A managed wrapper up class for the native c style peer connection APIs.
+  public class PeerConnectionM {
+    private const string dllPath = "webrtc_unity_plugin";
+
+    //create a peerconnection with turn servers
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern int CreatePeerConnection(string[] turnUrls, int noOfUrls,
+        string username, string credential);
+
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool ClosePeerConnection(int peerConnectionId);
+
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool AddStream(int peerConnectionId, bool audioOnly);
+
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool AddDataChannel(int peerConnectionId);
+
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool CreateOffer(int peerConnectionId);
+
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool CreateAnswer(int peerConnectionId);
+
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool SendDataViaDataChannel(int peerConnectionId, string data);
+
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool SetAudioControl(int peerConnectionId, bool isMute, bool isRecord);
+
+    [UnmanagedFunctionPointer(CallingConvention.Cdecl)]
+    private delegate void LocalDataChannelReadyInternalDelegate();
+    public delegate void LocalDataChannelReadyDelegate(int id);
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool RegisterOnLocalDataChannelReady(
+        int peerConnectionId, LocalDataChannelReadyInternalDelegate callback);
+
+    [UnmanagedFunctionPointer(CallingConvention.Cdecl)]
+    private delegate void DataFromDataChannelReadyInternalDelegate(string s);
+    public delegate void DataFromDataChannelReadyDelegate(int id, string s);
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool RegisterOnDataFromDataChannelReady(
+        int peerConnectionId, DataFromDataChannelReadyInternalDelegate callback);
+
+    [UnmanagedFunctionPointer(CallingConvention.Cdecl)]
+    private delegate void FailureMessageInternalDelegate(string msg);
+    public delegate void FailureMessageDelegate(int id, string msg);
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool RegisterOnFailure(int peerConnectionId,
+        FailureMessageInternalDelegate callback);
+
+    [UnmanagedFunctionPointer(CallingConvention.Cdecl)]
+    private delegate void AudioBusReadyInternalDelegate(IntPtr data, int bitsPerSample,
+        int sampleRate, int numberOfChannels, int numberOfFrames);
+    public delegate void AudioBusReadyDelegate(int id, IntPtr data, int bitsPerSample,
+        int sampleRate, int numberOfChannels, int numberOfFrames);
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool RegisterOnAudioBusReady(int peerConnectionId,
+        AudioBusReadyInternalDelegate callback);
+
+    // Video callbacks.
+    [UnmanagedFunctionPointer(CallingConvention.Cdecl)]
+    private delegate void I420FrameReadyInternalDelegate(
+        IntPtr dataY, IntPtr dataU, IntPtr dataV,
+        int strideY, int strideU, int strideV,
+        uint width, uint height);
+    public delegate void I420FrameReadyDelegate(int id,
+        IntPtr dataY, IntPtr dataU, IntPtr dataV,
+        int strideY, int strideU, int strideV,
+        uint width, uint height);
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool RegisterOnLocalI420FrameReady(int peerConnectionId,
+        I420FrameReadyInternalDelegate callback);
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool RegisterOnRemoteI420FrameReady(int peerConnectionId,
+        I420FrameReadyInternalDelegate callback);
+
+    [UnmanagedFunctionPointer(CallingConvention.Cdecl)]
+    private delegate void LocalSdpReadytoSendInternalDelegate(string type, string sdp);
+    public delegate void LocalSdpReadytoSendDelegate(int id, string type, string sdp);
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool RegisterOnLocalSdpReadytoSend(int peerConnectionId,
+        LocalSdpReadytoSendInternalDelegate callback);
+
+    [UnmanagedFunctionPointer(CallingConvention.Cdecl)]
+    private delegate void IceCandidateReadytoSendInternalDelegate(
+        string candidate, int sdpMlineIndex, string sdpMid);
+    public delegate void IceCandidateReadytoSendDelegate(
+        int id, string candidate, int sdpMlineIndex, string sdpMid);
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool RegisterOnIceCandidateReadytoSend(
+        int peerConnectionId, IceCandidateReadytoSendInternalDelegate callback);
+
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool SetRemoteDescription(int peerConnectionId, string type, string sdp);
+
+    [DllImport(dllPath, CallingConvention = CallingConvention.Cdecl)]
+    private static extern bool AddIceCandidate(int peerConnectionId, string sdp,
+      int sdpMlineindex, string sdpMid);
+
+    public PeerConnectionM(List<string> turnUrls, string username, string credential) {
+      string[] urls = turnUrls != null ? turnUrls.ToArray() : null;
+      int length = turnUrls != null ? turnUrls.Count : 0;
+      mPeerConnectionId = CreatePeerConnection(urls, length, username, credential);
+      RegisterCallbacks();
+    }
+
+    public void ClosePeerConnection() {
+      ClosePeerConnection(mPeerConnectionId);
+      mPeerConnectionId = -1;
+    }
+
+    // Return -1 if Peerconnection is not available.
+    public int GetUniqueId() {
+      return mPeerConnectionId;
+    }
+
+    public void AddStream(bool audioOnly) {
+      AddStream(mPeerConnectionId, audioOnly);
+    }
+
+    public void AddDataChannel() {
+      AddDataChannel(mPeerConnectionId);
+    }
+
+    public void CreateOffer() {
+      CreateOffer(mPeerConnectionId);
+    }
+
+    public void CreateAnswer() {
+      CreateAnswer(mPeerConnectionId);
+    }
+
+    public void SendDataViaDataChannel(string data) {
+      SendDataViaDataChannel(mPeerConnectionId, data);
+    }
+
+    public void SetAudioControl(bool isMute, bool isRecord) {
+      SetAudioControl(mPeerConnectionId, isMute, isRecord);
+    }
+
+    public void SetRemoteDescription(string type, string sdp) {
+      SetRemoteDescription(mPeerConnectionId, type, sdp);
+    }
+
+    public void AddIceCandidate(string candidate, int sdpMlineindex, string sdpMid) {
+      AddIceCandidate(mPeerConnectionId, candidate, sdpMlineindex, sdpMid);
+    }
+
+    private void RegisterCallbacks() {
+      localDataChannelReadyDelegate = new LocalDataChannelReadyInternalDelegate(
+          RaiseLocalDataChannelReady);
+      RegisterOnLocalDataChannelReady(mPeerConnectionId, localDataChannelReadyDelegate);
+
+      dataFromDataChannelReadyDelegate = new DataFromDataChannelReadyInternalDelegate(
+          RaiseDataFromDataChannelReady);
+      RegisterOnDataFromDataChannelReady(mPeerConnectionId, dataFromDataChannelReadyDelegate);
+
+      failureMessageDelegate = new FailureMessageInternalDelegate(RaiseFailureMessage);
+      RegisterOnFailure(mPeerConnectionId, failureMessageDelegate);
+
+      audioBusReadyDelegate = new AudioBusReadyInternalDelegate(RaiseAudioBusReady);
+      RegisterOnAudioBusReady(mPeerConnectionId, audioBusReadyDelegate);
+
+      localI420FrameReadyDelegate = new I420FrameReadyInternalDelegate(
+        RaiseLocalVideoFrameReady);
+      RegisterOnLocalI420FrameReady(mPeerConnectionId, localI420FrameReadyDelegate);
+
+      remoteI420FrameReadyDelegate = new I420FrameReadyInternalDelegate(
+        RaiseRemoteVideoFrameReady);
+      RegisterOnRemoteI420FrameReady(mPeerConnectionId, remoteI420FrameReadyDelegate);
+
+      localSdpReadytoSendDelegate = new LocalSdpReadytoSendInternalDelegate(
+        RaiseLocalSdpReadytoSend);
+      RegisterOnLocalSdpReadytoSend(mPeerConnectionId, localSdpReadytoSendDelegate);
+
+      iceCandidateReadytoSendDelegate =
+          new IceCandidateReadytoSendInternalDelegate(RaiseIceCandidateReadytoSend);
+      RegisterOnIceCandidateReadytoSend(
+          mPeerConnectionId, iceCandidateReadytoSendDelegate);
+    }
+
+    private void RaiseLocalDataChannelReady() {
+      if (OnLocalDataChannelReady != null)
+        OnLocalDataChannelReady(mPeerConnectionId);
+    }
+
+    private void RaiseDataFromDataChannelReady(string data) {
+      if (OnDataFromDataChannelReady != null)
+        OnDataFromDataChannelReady(mPeerConnectionId, data);
+    }
+
+    private void RaiseFailureMessage(string msg) {
+      if (OnFailureMessage != null)
+        OnFailureMessage(mPeerConnectionId, msg);
+    }
+
+    private void RaiseAudioBusReady(IntPtr data, int bitsPerSample,
+      int sampleRate, int numberOfChannels, int numberOfFrames) {
+      if (OnAudioBusReady != null)
+        OnAudioBusReady(mPeerConnectionId, data, bitsPerSample, sampleRate,
+            numberOfChannels, numberOfFrames);
+    }
+
+    private void RaiseLocalVideoFrameReady(
+        IntPtr dataY, IntPtr dataU, IntPtr dataV,
+        int strideY, int strideU, int strideV,
+        uint width, uint height) {
+      if (OnLocalVideoFrameReady != null)
+        OnLocalVideoFrameReady(mPeerConnectionId, dataY, dataU, dataV, strideY, strideU, strideV,
+          width, height);
+    }
+
+    private void RaiseRemoteVideoFrameReady(
+       IntPtr dataY, IntPtr dataU, IntPtr dataV,
+       int strideY, int strideU, int strideV,
+       uint width, uint height) {
+      if (OnRemoteVideoFrameReady != null)
+        OnRemoteVideoFrameReady(mPeerConnectionId, dataY, dataU, dataV, strideY, strideU, strideV,
+          width, height);
+    }
+
+
+    private void RaiseLocalSdpReadytoSend(string type, string sdp) {
+      if (OnLocalSdpReadytoSend != null)
+        OnLocalSdpReadytoSend(mPeerConnectionId, type, sdp);
+    }
+
+    private void RaiseIceCandidateReadytoSend(string candidate, int sdpMlineIndex, string sdpMid) {
+      if (OnIceCandidateReadytoSend != null)
+        OnIceCandidateReadytoSend(mPeerConnectionId, candidate, sdpMlineIndex, sdpMid);
+    }
+
+    public void AddQueuedIceCandidate(List<IceCandidate> iceCandidateQueue) {
+      if (iceCandidateQueue != null) {
+        foreach (IceCandidate ic in iceCandidateQueue) {
+          AddIceCandidate(mPeerConnectionId, ic.Candidate, ic.SdpMlineIndex, ic.SdpMid);
+        }
+      }
+    }
+
+    private LocalDataChannelReadyInternalDelegate localDataChannelReadyDelegate = null;
+    public event LocalDataChannelReadyDelegate OnLocalDataChannelReady;
+
+    private DataFromDataChannelReadyInternalDelegate dataFromDataChannelReadyDelegate = null;
+    public event DataFromDataChannelReadyDelegate OnDataFromDataChannelReady;
+
+    private FailureMessageInternalDelegate failureMessageDelegate = null;
+    public event FailureMessageDelegate OnFailureMessage;
+
+    private AudioBusReadyInternalDelegate audioBusReadyDelegate = null;
+    public event AudioBusReadyDelegate OnAudioBusReady;
+
+    private I420FrameReadyInternalDelegate localI420FrameReadyDelegate = null;
+    public event I420FrameReadyDelegate OnLocalVideoFrameReady;
+
+    private I420FrameReadyInternalDelegate remoteI420FrameReadyDelegate = null;
+    public event I420FrameReadyDelegate OnRemoteVideoFrameReady;
+
+    private LocalSdpReadytoSendInternalDelegate localSdpReadytoSendDelegate = null;
+    public event LocalSdpReadytoSendDelegate OnLocalSdpReadytoSend;
+
+    private IceCandidateReadytoSendInternalDelegate iceCandidateReadytoSendDelegate = null;
+    public event IceCandidateReadytoSendDelegate OnIceCandidateReadytoSend;
+
+    private int mPeerConnectionId = -1;
+  }
+}
diff --git a/third_party/libwebrtc/examples/unityplugin/class_reference_holder.cc b/third_party/libwebrtc/examples/unityplugin/class_reference_holder.cc
new file mode 100644
index 0000000000..00ca772e76
--- /dev/null
+++ b/third_party/libwebrtc/examples/unityplugin/class_reference_holder.cc
@@ -0,0 +1,88 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "examples/unityplugin/class_reference_holder.h"
+
+#include <utility>
+
+#include "sdk/android/src/jni/jni_helpers.h"
+
+namespace unity_plugin {
+
+// ClassReferenceHolder holds global reference to Java classes in app/webrtc.
+class ClassReferenceHolder {
+ public:
+  explicit ClassReferenceHolder(JNIEnv* jni);
+  ~ClassReferenceHolder();
+
+  void FreeReferences(JNIEnv* jni);
+  jclass GetClass(const std::string& name);
+
+  void LoadClass(JNIEnv* jni, const std::string& name);
+
+ private:
+  std::map<std::string, jclass> classes_;
+};
+
+// Allocated in LoadGlobalClassReferenceHolder(),
+// freed in FreeGlobalClassReferenceHolder().
+static ClassReferenceHolder* g_class_reference_holder = nullptr;
+
+void LoadGlobalClassReferenceHolder() {
+  RTC_CHECK(g_class_reference_holder == nullptr);
+  g_class_reference_holder = new ClassReferenceHolder(webrtc::jni::GetEnv());
+}
+
+void FreeGlobalClassReferenceHolder() {
+  g_class_reference_holder->FreeReferences(
+      webrtc::jni::AttachCurrentThreadIfNeeded());
+  delete g_class_reference_holder;
+  g_class_reference_holder = nullptr;
+}
+
+ClassReferenceHolder::ClassReferenceHolder(JNIEnv* jni) {
+  LoadClass(jni, "org/webrtc/UnityUtility");
+}
+
+ClassReferenceHolder::~ClassReferenceHolder() {
+  RTC_CHECK(classes_.empty()) << "Must call FreeReferences() before dtor!";
+}
+
+void ClassReferenceHolder::FreeReferences(JNIEnv* jni) {
+  for (std::map<std::string, jclass>::const_iterator it = classes_.begin();
+       it != classes_.end(); ++it) {
+    jni->DeleteGlobalRef(it->second);
+  }
+  classes_.clear();
+}
+
+jclass ClassReferenceHolder::GetClass(const std::string& name) {
+  std::map<std::string, jclass>::iterator it = classes_.find(name);
+  RTC_CHECK(it != classes_.end()) << "Unexpected GetClass() call for: " << name;
+  return it->second;
+}
+
+void ClassReferenceHolder::LoadClass(JNIEnv* jni, const std::string& name) {
+  jclass localRef = jni->FindClass(name.c_str());
+  CHECK_EXCEPTION(jni) << "error during FindClass: " << name;
+  RTC_CHECK(localRef) << name;
+  jclass globalRef = reinterpret_cast<jclass>(jni->NewGlobalRef(localRef));
+  CHECK_EXCEPTION(jni) << "error during NewGlobalRef: " << name;
+  RTC_CHECK(globalRef) << name;
+  bool inserted = classes_.insert(std::make_pair(name, globalRef)).second;
+  RTC_CHECK(inserted) << "Duplicate class name: " << name;
+}
+
+// Returns a global reference guaranteed to be valid for the lifetime of the
+// process.
+jclass FindClass(JNIEnv* jni, const char* name) {
+  return g_class_reference_holder->GetClass(name);
+}
+
+}  // namespace unity_plugin
diff --git a/third_party/libwebrtc/examples/unityplugin/class_reference_holder.h b/third_party/libwebrtc/examples/unityplugin/class_reference_holder.h
new file mode 100644
index 0000000000..884d471ceb
--- /dev/null
+++ b/third_party/libwebrtc/examples/unityplugin/class_reference_holder.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This is a supplement of webrtc::jni::ClassReferenceHolder.
+// The purpose of this ClassReferenceHolder is to load the example
+// specific java class into JNI c++ side, so that our c++ code can
+// call those java functions.
+
+#ifndef EXAMPLES_UNITYPLUGIN_CLASS_REFERENCE_HOLDER_H_
+#define EXAMPLES_UNITYPLUGIN_CLASS_REFERENCE_HOLDER_H_
+
+#include <jni.h>
+
+#include <map>
+#include <string>
+#include <vector>
+
+namespace unity_plugin {
+
+// LoadGlobalClassReferenceHolder must be called in JNI_OnLoad.
+void LoadGlobalClassReferenceHolder();
+// FreeGlobalClassReferenceHolder must be called in JNI_UnLoad.
+void FreeGlobalClassReferenceHolder();
+
+// Returns a global reference guaranteed to be valid for the lifetime of the
+// process.
+jclass FindClass(JNIEnv* jni, const char* name);
+
+}  // namespace unity_plugin
+
+#endif  // EXAMPLES_UNITYPLUGIN_CLASS_REFERENCE_HOLDER_H_
diff --git a/third_party/libwebrtc/examples/unityplugin/java/src/org/webrtc/UnityUtility.java b/third_party/libwebrtc/examples/unityplugin/java/src/org/webrtc/UnityUtility.java
new file mode 100644
index 0000000000..bd8bbfa449
--- /dev/null
+++ b/third_party/libwebrtc/examples/unityplugin/java/src/org/webrtc/UnityUtility.java
@@ -0,0 +1,68 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.content.Context;
+import androidx.annotation.Nullable;
+
+public class UnityUtility {
+  private static final String VIDEO_CAPTURER_THREAD_NAME = "VideoCapturerThread";
+
+  public static SurfaceTextureHelper LoadSurfaceTextureHelper() {
+    final SurfaceTextureHelper surfaceTextureHelper =
+        SurfaceTextureHelper.create(VIDEO_CAPTURER_THREAD_NAME, null);
+    return surfaceTextureHelper;
+  }
+
+  private static boolean useCamera2() {
+    return Camera2Enumerator.isSupported(ContextUtils.getApplicationContext());
+  }
+
+  private static @Nullable VideoCapturer createCameraCapturer(CameraEnumerator enumerator) {
+    final String[] deviceNames = enumerator.getDeviceNames();
+
+    for (String deviceName : deviceNames) {
+      if (enumerator.isFrontFacing(deviceName)) {
+        VideoCapturer videoCapturer = enumerator.createCapturer(deviceName, null);
+
+        if (videoCapturer != null) {
+          return videoCapturer;
+        }
+      }
+    }
+
+    return null;
+  }
+
+  public static VideoCapturer LinkCamera(
+      long nativeTrackSource, SurfaceTextureHelper surfaceTextureHelper) {
+    VideoCapturer capturer =
+        createCameraCapturer(new Camera2Enumerator(ContextUtils.getApplicationContext()));
+
+    VideoSource videoSource = new VideoSource(nativeTrackSource);
+
+    capturer.initialize(surfaceTextureHelper, ContextUtils.getApplicationContext(),
+        videoSource.getCapturerObserver());
+
+    capturer.startCapture(720, 480, 30);
+    return capturer;
+  }
+
+  public static void StopCamera(VideoCapturer camera) throws InterruptedException {
+    camera.stopCapture();
+    camera.dispose();
+  }
+
+  public static void InitializePeerConncectionFactory(Context context) throws InterruptedException {
+    PeerConnectionFactory.initialize(
+        PeerConnectionFactory.InitializationOptions.builder(context).createInitializationOptions());
+  }
+}
diff --git a/third_party/libwebrtc/examples/unityplugin/jni_onload.cc b/third_party/libwebrtc/examples/unityplugin/jni_onload.cc
new file mode 100644
index 0000000000..b9c92d5ef4
--- /dev/null
+++ b/third_party/libwebrtc/examples/unityplugin/jni_onload.cc
@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <jni.h>
+#undef JNIEXPORT
+#define JNIEXPORT __attribute__((visibility("default")))
+
+#include "examples/unityplugin/class_reference_holder.h"
+#include "rtc_base/ssl_adapter.h"
+#include "sdk/android/native_api/jni/class_loader.h"
+#include "sdk/android/src/jni/jni_helpers.h"
+
+namespace webrtc {
+namespace jni {
+
+extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM* jvm, void* reserved) {
+  jint ret = InitGlobalJniVariables(jvm);
+  RTC_DCHECK_GE(ret, 0);
+  if (ret < 0)
+    return -1;
+
+  RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
+  webrtc::InitClassLoader(GetEnv());
+  unity_plugin::LoadGlobalClassReferenceHolder();
+
+  return ret;
+}
+
+extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM* jvm, void* reserved) {
+  unity_plugin::FreeGlobalClassReferenceHolder();
+  RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()";
+}
+
+}  // namespace jni
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/examples/unityplugin/simple_peer_connection.cc b/third_party/libwebrtc/examples/unityplugin/simple_peer_connection.cc
new file mode 100644
index 0000000000..861b22f29c
--- /dev/null
+++ b/third_party/libwebrtc/examples/unityplugin/simple_peer_connection.cc
@@ -0,0 +1,587 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/unityplugin/simple_peer_connection.h"
+
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/create_peerconnection_factory.h"
+#include "media/engine/internal_decoder_factory.h"
+#include "media/engine/internal_encoder_factory.h"
+#include "media/engine/multiplex_codec_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/video_capture/video_capture_factory.h"
+#include "pc/video_track_source.h"
+#include "test/vcm_capturer.h"
+
+#if defined(WEBRTC_ANDROID)
+#include "examples/unityplugin/class_reference_holder.h"
+#include "modules/utility/include/helpers_android.h"
+#include "sdk/android/src/jni/android_video_track_source.h"
+#include "sdk/android/src/jni/jni_helpers.h"
+#endif
+
+// Names used for media stream ids.
+const char kAudioLabel[] = "audio_label";
+const char kVideoLabel[] = "video_label";
+const char kStreamId[] = "stream_id";
+
+namespace {
+static int g_peer_count = 0;
+static std::unique_ptr<rtc::Thread> g_worker_thread;
+static std::unique_ptr<rtc::Thread> g_signaling_thread;
+static rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
+    g_peer_connection_factory;
+#if defined(WEBRTC_ANDROID)
+// Android case: the video track does not own the capturer, and it
+// relies on the app to dispose the capturer when the peerconnection
+// shuts down.
+static jobject g_camera = nullptr;
+#else
+class CapturerTrackSource : public webrtc::VideoTrackSource {
+ public:
+  static rtc::scoped_refptr<CapturerTrackSource> Create() {
+    const size_t kWidth = 640;
+    const size_t kHeight = 480;
+    const size_t kFps = 30;
+    const size_t kDeviceIndex = 0;
+    std::unique_ptr<webrtc::test::VcmCapturer> capturer = absl::WrapUnique(
+        webrtc::test::VcmCapturer::Create(kWidth, kHeight, kFps, kDeviceIndex));
+    if (!capturer) {
+      return nullptr;
+    }
+    return rtc::make_ref_counted<CapturerTrackSource>(std::move(capturer));
+  }
+
+ protected:
+  explicit CapturerTrackSource(
+      std::unique_ptr<webrtc::test::VcmCapturer> capturer)
+      : VideoTrackSource(/*remote=*/false), capturer_(std::move(capturer)) {}
+
+ private:
+  rtc::VideoSourceInterface<webrtc::VideoFrame>* source() override {
+    return capturer_.get();
+  }
+  std::unique_ptr<webrtc::test::VcmCapturer> capturer_;
+};
+
+#endif
+
+std::string GetEnvVarOrDefault(const char* env_var_name,
+                               const char* default_value) {
+  std::string value;
+  const char* env_var = getenv(env_var_name);
+  if (env_var)
+    value = env_var;
+
+  if (value.empty())
+    value = default_value;
+
+  return value;
+}
+
+std::string GetPeerConnectionString() {
+  return GetEnvVarOrDefault("WEBRTC_CONNECT", "stun:stun.l.google.com:19302");
+}
+
+class DummySetSessionDescriptionObserver
+    : public webrtc::SetSessionDescriptionObserver {
+ public:
+  static rtc::scoped_refptr<DummySetSessionDescriptionObserver> Create() {
+    return rtc::make_ref_counted<DummySetSessionDescriptionObserver>();
+  }
+  virtual void OnSuccess() { RTC_LOG(LS_INFO) << __FUNCTION__; }
+  virtual void OnFailure(webrtc::RTCError error) {
+    RTC_LOG(LS_INFO) << __FUNCTION__ << " " << ToString(error.type()) << ": "
+                     << error.message();
+  }
+
+ protected:
+  DummySetSessionDescriptionObserver() {}
+  ~DummySetSessionDescriptionObserver() {}
+};
+
+}  // namespace
+
+bool SimplePeerConnection::InitializePeerConnection(const char** turn_urls,
+                                                    const int no_of_urls,
+                                                    const char* username,
+                                                    const char* credential,
+                                                    bool is_receiver) {
+  RTC_DCHECK(peer_connection_.get() == nullptr);
+
+  if (g_peer_connection_factory == nullptr) {
+    g_worker_thread = rtc::Thread::Create();
+    g_worker_thread->Start();
+    g_signaling_thread = rtc::Thread::Create();
+    g_signaling_thread->Start();
+
+    g_peer_connection_factory = webrtc::CreatePeerConnectionFactory(
+        g_worker_thread.get(), g_worker_thread.get(), g_signaling_thread.get(),
+        nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+        webrtc::CreateBuiltinAudioDecoderFactory(),
+        std::unique_ptr<webrtc::VideoEncoderFactory>(
+            new webrtc::MultiplexEncoderFactory(
+                std::make_unique<webrtc::InternalEncoderFactory>())),
+        std::unique_ptr<webrtc::VideoDecoderFactory>(
+            new webrtc::MultiplexDecoderFactory(
+                std::make_unique<webrtc::InternalDecoderFactory>())),
+        nullptr, nullptr);
+  }
+  if (!g_peer_connection_factory.get()) {
+    DeletePeerConnection();
+    return false;
+  }
+
+  g_peer_count++;
+  if (!CreatePeerConnection(turn_urls, no_of_urls, username, credential)) {
+    DeletePeerConnection();
+    return false;
+  }
+
+  mandatory_receive_ = is_receiver;
+  return peer_connection_.get() != nullptr;
+}
+
+bool SimplePeerConnection::CreatePeerConnection(const char** turn_urls,
+                                                const int no_of_urls,
+                                                const char* username,
+                                                const char* credential) {
+  RTC_DCHECK(g_peer_connection_factory.get() != nullptr);
+  RTC_DCHECK(peer_connection_.get() == nullptr);
+
+  local_video_observer_.reset(new VideoObserver());
+  remote_video_observer_.reset(new VideoObserver());
+
+  // Add the turn server.
+  if (turn_urls != nullptr) {
+    if (no_of_urls > 0) {
+      webrtc::PeerConnectionInterface::IceServer turn_server;
+      for (int i = 0; i < no_of_urls; i++) {
+        std::string url(turn_urls[i]);
+        if (url.length() > 0)
+          turn_server.urls.push_back(turn_urls[i]);
+      }
+
+      std::string user_name(username);
+      if (user_name.length() > 0)
+        turn_server.username = username;
+
+      std::string password(credential);
+      if (password.length() > 0)
+        turn_server.password = credential;
+
+      config_.servers.push_back(turn_server);
+    }
+  }
+
+  // Add the stun server.
+  webrtc::PeerConnectionInterface::IceServer stun_server;
+  stun_server.uri = GetPeerConnectionString();
+  config_.servers.push_back(stun_server);
+
+  auto result = g_peer_connection_factory->CreatePeerConnectionOrError(
+      config_, webrtc::PeerConnectionDependencies(this));
+  if (!result.ok()) {
+    peer_connection_ = nullptr;
+    return false;
+  }
+  peer_connection_ = result.MoveValue();
+  return true;
+}
+
+void SimplePeerConnection::DeletePeerConnection() {
+  g_peer_count--;
+
+#if defined(WEBRTC_ANDROID)
+  if (g_camera) {
+    JNIEnv* env = webrtc::jni::GetEnv();
+    jclass pc_factory_class =
+        unity_plugin::FindClass(env, "org/webrtc/UnityUtility");
+    jmethodID stop_camera_method = webrtc::GetStaticMethodID(
+        env, pc_factory_class, "StopCamera", "(Lorg/webrtc/VideoCapturer;)V");
+
+    env->CallStaticVoidMethod(pc_factory_class, stop_camera_method, g_camera);
+    CHECK_EXCEPTION(env);
+
+    g_camera = nullptr;
+  }
+#endif
+
+  CloseDataChannel();
+  peer_connection_ = nullptr;
+  active_streams_.clear();
+
+  if (g_peer_count == 0) {
+    g_peer_connection_factory = nullptr;
+    g_signaling_thread.reset();
+    g_worker_thread.reset();
+  }
+}
+
+bool SimplePeerConnection::CreateOffer() {
+  if (!peer_connection_.get())
+    return false;
+
+  webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
+  if (mandatory_receive_) {
+    options.offer_to_receive_audio = true;
+    options.offer_to_receive_video = true;
+  }
+  peer_connection_->CreateOffer(this, options);
+  return true;
+}
+
+bool SimplePeerConnection::CreateAnswer() {
+  if (!peer_connection_.get())
+    return false;
+
+  webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
+  if (mandatory_receive_) {
+    options.offer_to_receive_audio = true;
+    options.offer_to_receive_video = true;
+  }
+  peer_connection_->CreateAnswer(this, options);
+  return true;
+}
+
+void SimplePeerConnection::OnSuccess(
+    webrtc::SessionDescriptionInterface* desc) {
+  peer_connection_->SetLocalDescription(
+      DummySetSessionDescriptionObserver::Create().get(), desc);
+
+  std::string sdp;
+  desc->ToString(&sdp);
+
+  if (OnLocalSdpReady)
+    OnLocalSdpReady(desc->type().c_str(), sdp.c_str());
+}
+
+void SimplePeerConnection::OnFailure(webrtc::RTCError error) {
+  RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message();
+
+  // TODO(hta): include error.type in the message
+  if (OnFailureMessage)
+    OnFailureMessage(error.message());
+}
+
+void SimplePeerConnection::OnIceCandidate(
+    const webrtc::IceCandidateInterface* candidate) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << " " << candidate->sdp_mline_index();
+
+  std::string sdp;
+  if (!candidate->ToString(&sdp)) {
+    RTC_LOG(LS_ERROR) << "Failed to serialize candidate";
+    return;
+  }
+
+  if (OnIceCandidateReady)
+    OnIceCandidateReady(sdp.c_str(), candidate->sdp_mline_index(),
+                        candidate->sdp_mid().c_str());
+}
+
+void SimplePeerConnection::RegisterOnLocalI420FrameReady(
+    I420FRAMEREADY_CALLBACK callback) {
+  if (local_video_observer_)
+    local_video_observer_->SetVideoCallback(callback);
+}
+
+void SimplePeerConnection::RegisterOnRemoteI420FrameReady(
+    I420FRAMEREADY_CALLBACK callback) {
+  if (remote_video_observer_)
+    remote_video_observer_->SetVideoCallback(callback);
+}
+
+void SimplePeerConnection::RegisterOnLocalDataChannelReady(
+    LOCALDATACHANNELREADY_CALLBACK callback) {
+  OnLocalDataChannelReady = callback;
+}
+
+void SimplePeerConnection::RegisterOnDataFromDataChannelReady(
+    DATAFROMEDATECHANNELREADY_CALLBACK callback) {
+  OnDataFromDataChannelReady = callback;
+}
+
+void SimplePeerConnection::RegisterOnFailure(FAILURE_CALLBACK callback) {
+  OnFailureMessage = callback;
+}
+
+void SimplePeerConnection::RegisterOnAudioBusReady(
+    AUDIOBUSREADY_CALLBACK callback) {
+  OnAudioReady = callback;
+}
+
+void SimplePeerConnection::RegisterOnLocalSdpReadytoSend(
+    LOCALSDPREADYTOSEND_CALLBACK callback) {
+  OnLocalSdpReady = callback;
+}
+
+void SimplePeerConnection::RegisterOnIceCandidateReadytoSend(
+    ICECANDIDATEREADYTOSEND_CALLBACK callback) {
+  OnIceCandidateReady = callback;
+}
+
+bool SimplePeerConnection::SetRemoteDescription(const char* type,
+                                                const char* sdp) {
+  if (!peer_connection_)
+    return false;
+
+  std::string remote_desc(sdp);
+  std::string desc_type(type);
+  webrtc::SdpParseError error;
+  webrtc::SessionDescriptionInterface* session_description(
+      webrtc::CreateSessionDescription(desc_type, remote_desc, &error));
+  if (!session_description) {
+    RTC_LOG(LS_WARNING) << "Can't parse received session description message. "
+                           "SdpParseError was: "
+                        << error.description;
+    return false;
+  }
+  RTC_LOG(LS_INFO) << " Received session description :" << remote_desc;
+  peer_connection_->SetRemoteDescription(
+      DummySetSessionDescriptionObserver::Create().get(), session_description);
+
+  return true;
+}
+
+bool SimplePeerConnection::AddIceCandidate(const char* candidate,
+                                           const int sdp_mlineindex,
+                                           const char* sdp_mid) {
+  if (!peer_connection_)
+    return false;
+
+  webrtc::SdpParseError error;
+  std::unique_ptr<webrtc::IceCandidateInterface> ice_candidate(
+      webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, candidate, &error));
+  if (!ice_candidate.get()) {
+    RTC_LOG(LS_WARNING) << "Can't parse received candidate message. "
+                           "SdpParseError was: "
+                        << error.description;
+    return false;
+  }
+  if (!peer_connection_->AddIceCandidate(ice_candidate.get())) {
+    RTC_LOG(LS_WARNING) << "Failed to apply the received candidate";
+    return false;
+  }
+  RTC_LOG(LS_INFO) << " Received candidate :" << candidate;
+  return true;
+}
+
+void SimplePeerConnection::SetAudioControl(bool is_mute, bool is_record) {
+  is_mute_audio_ = is_mute;
+  is_record_audio_ = is_record;
+
+  SetAudioControl();
+}
+
+void SimplePeerConnection::SetAudioControl() {
+  if (!remote_stream_)
+    return;
+  webrtc::AudioTrackVector tracks = remote_stream_->GetAudioTracks();
+  if (tracks.empty())
+    return;
+
+  rtc::scoped_refptr<webrtc::AudioTrackInterface>& audio_track = tracks[0];
+  if (is_record_audio_)
+    audio_track->AddSink(this);
+  else
+    audio_track->RemoveSink(this);
+
+  for (auto& track : tracks) {
+    if (is_mute_audio_)
+      track->set_enabled(false);
+    else
+      track->set_enabled(true);
+  }
+}
+
+void SimplePeerConnection::OnAddStream(
+    rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << " " << stream->id();
+  remote_stream_ = stream;
+  if (remote_video_observer_ && !remote_stream_->GetVideoTracks().empty()) {
+    remote_stream_->GetVideoTracks()[0]->AddOrUpdateSink(
+        remote_video_observer_.get(), rtc::VideoSinkWants());
+  }
+  SetAudioControl();
+}
+
+void SimplePeerConnection::AddStreams(bool audio_only) {
+  if (active_streams_.find(kStreamId) != active_streams_.end())
+    return;  // Already added.
+
+  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+      g_peer_connection_factory->CreateLocalMediaStream(kStreamId);
+
+  rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+      g_peer_connection_factory->CreateAudioTrack(
+          kAudioLabel,
+          g_peer_connection_factory->CreateAudioSource(cricket::AudioOptions())
+              .get()));
+  stream->AddTrack(audio_track);
+
+  if (!audio_only) {
+#if defined(WEBRTC_ANDROID)
+    JNIEnv* env = webrtc::jni::GetEnv();
+    jclass pc_factory_class =
+        unity_plugin::FindClass(env, "org/webrtc/UnityUtility");
+    jmethodID load_texture_helper_method = webrtc::GetStaticMethodID(
+        env, pc_factory_class, "LoadSurfaceTextureHelper",
+        "()Lorg/webrtc/SurfaceTextureHelper;");
+    jobject texture_helper = env->CallStaticObjectMethod(
+        pc_factory_class, load_texture_helper_method);
+    CHECK_EXCEPTION(env);
+    RTC_DCHECK(texture_helper != nullptr)
+        << "Cannot get the Surface Texture Helper.";
+
+    auto source = rtc::make_ref_counted<webrtc::jni::AndroidVideoTrackSource>(
+        g_signaling_thread.get(), env, /*is_screencast=*/false,
+        /*align_timestamps=*/true);
+
+    // link with VideoCapturer (Camera);
+    jmethodID link_camera_method = webrtc::GetStaticMethodID(
+        env, pc_factory_class, "LinkCamera",
+        "(JLorg/webrtc/SurfaceTextureHelper;)Lorg/webrtc/VideoCapturer;");
+    jobject camera_tmp =
+        env->CallStaticObjectMethod(pc_factory_class, link_camera_method,
+                                    (jlong)source.get(), texture_helper);
+    CHECK_EXCEPTION(env);
+    g_camera = (jobject)env->NewGlobalRef(camera_tmp);
+
+    rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
+        g_peer_connection_factory->CreateVideoTrack(kVideoLabel,
+                                                    source.release()));
+    stream->AddTrack(video_track);
+#else
+    rtc::scoped_refptr<CapturerTrackSource> video_device =
+        CapturerTrackSource::Create();
+    if (video_device) {
+      rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
+          g_peer_connection_factory->CreateVideoTrack(kVideoLabel,
+                                                      video_device.get()));
+
+      stream->AddTrack(video_track);
+    }
+#endif
+    if (local_video_observer_ && !stream->GetVideoTracks().empty()) {
+      stream->GetVideoTracks()[0]->AddOrUpdateSink(local_video_observer_.get(),
+                                                   rtc::VideoSinkWants());
+    }
+  }
+
+  if (!peer_connection_->AddStream(stream.get())) {
+    RTC_LOG(LS_ERROR) << "Adding stream to PeerConnection failed";
+  }
+
+  typedef std::pair<std::string,
+                    rtc::scoped_refptr<webrtc::MediaStreamInterface>>
+      MediaStreamPair;
+  active_streams_.insert(MediaStreamPair(stream->id(), stream));
+}
+
+bool SimplePeerConnection::CreateDataChannel() {
+  struct webrtc::DataChannelInit init;
+  init.ordered = true;
+  init.reliable = true;
+  auto result = peer_connection_->CreateDataChannelOrError("Hello", &init);
+  if (result.ok()) {
+    data_channel_ = result.MoveValue();
+    data_channel_->RegisterObserver(this);
+    RTC_LOG(LS_INFO) << "Succeeds to create data channel";
+    return true;
+  } else {
+    RTC_LOG(LS_INFO) << "Fails to create data channel";
+    return false;
+  }
+}
+
+void SimplePeerConnection::CloseDataChannel() {
+  if (data_channel_.get()) {
+    data_channel_->UnregisterObserver();
+    data_channel_->Close();
+  }
+  data_channel_ = nullptr;
+}
+
+bool SimplePeerConnection::SendDataViaDataChannel(const std::string& data) {
+  if (!data_channel_.get()) {
+    RTC_LOG(LS_INFO) << "Data channel is not established";
+    return false;
+  }
+  webrtc::DataBuffer buffer(data);
+  data_channel_->Send(buffer);
+  return true;
+}
+
+// Peerconnection observer
+void SimplePeerConnection::OnDataChannel(
+    rtc::scoped_refptr<webrtc::DataChannelInterface> channel) {
+  channel->RegisterObserver(this);
+}
+
+void SimplePeerConnection::OnStateChange() {
+  if (data_channel_) {
+    webrtc::DataChannelInterface::DataState state = data_channel_->state();
+    if (state == webrtc::DataChannelInterface::kOpen) {
+      if (OnLocalDataChannelReady)
+        OnLocalDataChannelReady();
+      RTC_LOG(LS_INFO) << "Data channel is open";
+    }
+  }
+}
+
+//  A data buffer was successfully received.
+void SimplePeerConnection::OnMessage(const webrtc::DataBuffer& buffer) {
+  size_t size = buffer.data.size();
+  char* msg = new char[size + 1];
+  memcpy(msg, buffer.data.data(), size);
+  msg[size] = 0;
+  if (OnDataFromDataChannelReady)
+    OnDataFromDataChannelReady(msg);
+  delete[] msg;
+}
+
+// AudioTrackSinkInterface implementation.
+void SimplePeerConnection::OnData(const void* audio_data,
+                                  int bits_per_sample,
+                                  int sample_rate,
+                                  size_t number_of_channels,
+                                  size_t number_of_frames) {
+  if (OnAudioReady)
+    OnAudioReady(audio_data, bits_per_sample, sample_rate,
+                 static_cast<int>(number_of_channels),
+                 static_cast<int>(number_of_frames));
+}
+
+std::vector<uint32_t> SimplePeerConnection::GetRemoteAudioTrackSsrcs() {
+  std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>> receivers =
+      peer_connection_->GetReceivers();
+
+  std::vector<uint32_t> ssrcs;
+  for (const auto& receiver : receivers) {
+    if (receiver->media_type() != cricket::MEDIA_TYPE_AUDIO)
+      continue;
+
+    std::vector<webrtc::RtpEncodingParameters> params =
+        receiver->GetParameters().encodings;
+
+    for (const auto& param : params) {
+      uint32_t ssrc = param.ssrc.value_or(0);
+      if (ssrc > 0)
+        ssrcs.push_back(ssrc);
+    }
+  }
+
+  return ssrcs;
+}
diff --git a/third_party/libwebrtc/examples/unityplugin/simple_peer_connection.h b/third_party/libwebrtc/examples/unityplugin/simple_peer_connection.h
new file mode 100644
index 0000000000..de652ef118
--- /dev/null
+++ b/third_party/libwebrtc/examples/unityplugin/simple_peer_connection.h
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_
+#define EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/data_channel_interface.h"
+#include "api/media_stream_interface.h"
+#include "api/peer_connection_interface.h"
+#include "examples/unityplugin/unity_plugin_apis.h"
+#include "examples/unityplugin/video_observer.h"
+
+class SimplePeerConnection : public webrtc::PeerConnectionObserver,
+                             public webrtc::CreateSessionDescriptionObserver,
+                             public webrtc::DataChannelObserver,
+                             public webrtc::AudioTrackSinkInterface {
+ public:
+  SimplePeerConnection() {}
+  ~SimplePeerConnection() {}
+
+  bool InitializePeerConnection(const char** turn_urls,
+                                int no_of_urls,
+                                const char* username,
+                                const char* credential,
+                                bool is_receiver);
+  void DeletePeerConnection();
+  void AddStreams(bool audio_only);
+  bool CreateDataChannel();
+  bool CreateOffer();
+  bool CreateAnswer();
+  bool SendDataViaDataChannel(const std::string& data);
+  void SetAudioControl(bool is_mute, bool is_record);
+
+  // Register callback functions.
+  void RegisterOnLocalI420FrameReady(I420FRAMEREADY_CALLBACK callback);
+  void RegisterOnRemoteI420FrameReady(I420FRAMEREADY_CALLBACK callback);
+  void RegisterOnLocalDataChannelReady(LOCALDATACHANNELREADY_CALLBACK callback);
+  void RegisterOnDataFromDataChannelReady(
+      DATAFROMEDATECHANNELREADY_CALLBACK callback);
+  void RegisterOnFailure(FAILURE_CALLBACK callback);
+  void RegisterOnAudioBusReady(AUDIOBUSREADY_CALLBACK callback);
+  void RegisterOnLocalSdpReadytoSend(LOCALSDPREADYTOSEND_CALLBACK callback);
+  void RegisterOnIceCandidateReadytoSend(
+      ICECANDIDATEREADYTOSEND_CALLBACK callback);
+  bool SetRemoteDescription(const char* type, const char* sdp);
+  bool AddIceCandidate(const char* sdp,
+                       int sdp_mlineindex,
+                       const char* sdp_mid);
+
+ protected:
+  // create a peerconneciton and add the turn servers info to the configuration.
+  bool CreatePeerConnection(const char** turn_urls,
+                            int no_of_urls,
+                            const char* username,
+                            const char* credential);
+  void CloseDataChannel();
+  void SetAudioControl();
+
+  // PeerConnectionObserver implementation.
+  void OnSignalingChange(
+      webrtc::PeerConnectionInterface::SignalingState new_state) override {}
+  void OnAddStream(
+      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override;
+  void OnRemoveStream(
+      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override {}
+  void OnDataChannel(
+      rtc::scoped_refptr<webrtc::DataChannelInterface> channel) override;
+  void OnRenegotiationNeeded() override {}
+  void OnIceConnectionChange(
+      webrtc::PeerConnectionInterface::IceConnectionState new_state) override {}
+  void OnIceGatheringChange(
+      webrtc::PeerConnectionInterface::IceGatheringState new_state) override {}
+  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
+  void OnIceConnectionReceivingChange(bool receiving) override {}
+
+  // CreateSessionDescriptionObserver implementation.
+  void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
+  void OnFailure(webrtc::RTCError error) override;
+
+  // DataChannelObserver implementation.
+  void OnStateChange() override;
+  void OnMessage(const webrtc::DataBuffer& buffer) override;
+
+  // AudioTrackSinkInterface implementation.
+  void OnData(const void* audio_data,
+              int bits_per_sample,
+              int sample_rate,
+              size_t number_of_channels,
+              size_t number_of_frames) override;
+
+  // Get remote audio tracks ssrcs.
+  std::vector<uint32_t> GetRemoteAudioTrackSsrcs();
+
+ private:
+  rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
+  rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel_;
+  std::map<std::string, rtc::scoped_refptr<webrtc::MediaStreamInterface> >
+      active_streams_;
+
+  std::unique_ptr<VideoObserver> local_video_observer_;
+  std::unique_ptr<VideoObserver> remote_video_observer_;
+
+  rtc::scoped_refptr<webrtc::MediaStreamInterface> remote_stream_ = nullptr;
+  webrtc::PeerConnectionInterface::RTCConfiguration config_;
+
+  LOCALDATACHANNELREADY_CALLBACK OnLocalDataChannelReady = nullptr;
+  DATAFROMEDATECHANNELREADY_CALLBACK OnDataFromDataChannelReady = nullptr;
+  FAILURE_CALLBACK OnFailureMessage = nullptr;
+  AUDIOBUSREADY_CALLBACK OnAudioReady = nullptr;
+
+  LOCALSDPREADYTOSEND_CALLBACK OnLocalSdpReady = nullptr;
+  ICECANDIDATEREADYTOSEND_CALLBACK OnIceCandidateReady = nullptr;
+
+  bool is_mute_audio_ = false;
+  bool is_record_audio_ = false;
+  bool mandatory_receive_ = false;
+
+  // disallow copy-and-assign
+  SimplePeerConnection(const SimplePeerConnection&) = delete;
+  SimplePeerConnection& operator=(const SimplePeerConnection&) = delete;
+};
+
+#endif  // EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_
diff --git a/third_party/libwebrtc/examples/unityplugin/unity_plugin_apis.cc b/third_party/libwebrtc/examples/unityplugin/unity_plugin_apis.cc
new file mode 100644
index 0000000000..6e34d7e1e0
--- /dev/null
+++ b/third_party/libwebrtc/examples/unityplugin/unity_plugin_apis.cc
@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/unityplugin/unity_plugin_apis.h"
+
+#include <map>
+#include <string>
+
+#include "examples/unityplugin/simple_peer_connection.h"
+
+namespace {
+static int g_peer_connection_id = 1;
+static std::map<int, rtc::scoped_refptr<SimplePeerConnection>>
+    g_peer_connection_map;
+}  // namespace
+
+int CreatePeerConnection(const char** turn_urls,
+                         const int no_of_urls,
+                         const char* username,
+                         const char* credential,
+                         bool mandatory_receive_video) {
+  g_peer_connection_map[g_peer_connection_id] =
+      rtc::make_ref_counted<SimplePeerConnection>();
+
+  if (!g_peer_connection_map[g_peer_connection_id]->InitializePeerConnection(
+          turn_urls, no_of_urls, username, credential, mandatory_receive_video))
+    return -1;
+
+  return g_peer_connection_id++;
+}
+
+bool ClosePeerConnection(int peer_connection_id) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  g_peer_connection_map[peer_connection_id]->DeletePeerConnection();
+  g_peer_connection_map.erase(peer_connection_id);
+  return true;
+}
+
+bool AddStream(int peer_connection_id, bool audio_only) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  g_peer_connection_map[peer_connection_id]->AddStreams(audio_only);
+  return true;
+}
+
+bool AddDataChannel(int peer_connection_id) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  return g_peer_connection_map[peer_connection_id]->CreateDataChannel();
+}
+
+bool CreateOffer(int peer_connection_id) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  return g_peer_connection_map[peer_connection_id]->CreateOffer();
+}
+
+bool CreateAnswer(int peer_connection_id) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  return g_peer_connection_map[peer_connection_id]->CreateAnswer();
+}
+
+bool SendDataViaDataChannel(int peer_connection_id, const char* data) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  std::string s(data);
+  g_peer_connection_map[peer_connection_id]->SendDataViaDataChannel(s);
+
+  return true;
+}
+
+bool SetAudioControl(int peer_connection_id, bool is_mute, bool is_record) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  g_peer_connection_map[peer_connection_id]->SetAudioControl(is_mute,
+                                                             is_record);
+  return true;
+}
+
+bool SetRemoteDescription(int peer_connection_id,
+                          const char* type,
+                          const char* sdp) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  return g_peer_connection_map[peer_connection_id]->SetRemoteDescription(type,
+                                                                         sdp);
+}
+
+bool AddIceCandidate(const int peer_connection_id,
+                     const char* candidate,
+                     const int sdp_mlineindex,
+                     const char* sdp_mid) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  return g_peer_connection_map[peer_connection_id]->AddIceCandidate(
+      candidate, sdp_mlineindex, sdp_mid);
+}
+
+// Register callback functions.
+bool RegisterOnLocalI420FrameReady(int peer_connection_id,
+                                   I420FRAMEREADY_CALLBACK callback) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  g_peer_connection_map[peer_connection_id]->RegisterOnLocalI420FrameReady(
+      callback);
+  return true;
+}
+
+bool RegisterOnRemoteI420FrameReady(int peer_connection_id,
+                                    I420FRAMEREADY_CALLBACK callback) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  g_peer_connection_map[peer_connection_id]->RegisterOnRemoteI420FrameReady(
+      callback);
+  return true;
+}
+
+bool RegisterOnLocalDataChannelReady(int peer_connection_id,
+                                     LOCALDATACHANNELREADY_CALLBACK callback) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  g_peer_connection_map[peer_connection_id]->RegisterOnLocalDataChannelReady(
+      callback);
+  return true;
+}
+
+bool RegisterOnDataFromDataChannelReady(
+    int peer_connection_id,
+    DATAFROMEDATECHANNELREADY_CALLBACK callback) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  g_peer_connection_map[peer_connection_id]->RegisterOnDataFromDataChannelReady(
+      callback);
+  return true;
+}
+
+bool RegisterOnFailure(int peer_connection_id, FAILURE_CALLBACK callback) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  g_peer_connection_map[peer_connection_id]->RegisterOnFailure(callback);
+  return true;
+}
+
+bool RegisterOnAudioBusReady(int peer_connection_id,
+                             AUDIOBUSREADY_CALLBACK callback) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  g_peer_connection_map[peer_connection_id]->RegisterOnAudioBusReady(callback);
+  return true;
+}
+
+// Singnaling channel related functions.
+bool RegisterOnLocalSdpReadytoSend(int peer_connection_id,
+                                   LOCALSDPREADYTOSEND_CALLBACK callback) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  g_peer_connection_map[peer_connection_id]->RegisterOnLocalSdpReadytoSend(
+      callback);
+  return true;
+}
+
+bool RegisterOnIceCandidateReadytoSend(
+    int peer_connection_id,
+    ICECANDIDATEREADYTOSEND_CALLBACK callback) {
+  if (!g_peer_connection_map.count(peer_connection_id))
+    return false;
+
+  g_peer_connection_map[peer_connection_id]->RegisterOnIceCandidateReadytoSend(
+      callback);
+  return true;
+}
diff --git a/third_party/libwebrtc/examples/unityplugin/unity_plugin_apis.h b/third_party/libwebrtc/examples/unityplugin/unity_plugin_apis.h
new file mode 100644
index 0000000000..9790dc57b9
--- /dev/null
+++ b/third_party/libwebrtc/examples/unityplugin/unity_plugin_apis.h
@@ -0,0 +1,108 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file provides an example of unity native plugin APIs.
+
+#ifndef EXAMPLES_UNITYPLUGIN_UNITY_PLUGIN_APIS_H_
+#define EXAMPLES_UNITYPLUGIN_UNITY_PLUGIN_APIS_H_
+
+#include <stdint.h>
+
+// Definitions of callback functions.
+typedef void (*I420FRAMEREADY_CALLBACK)(const uint8_t* data_y,
+                                        const uint8_t* data_u,
+                                        const uint8_t* data_v,
+                                        const uint8_t* data_a,
+                                        int stride_y,
+                                        int stride_u,
+                                        int stride_v,
+                                        int stride_a,
+                                        uint32_t width,
+                                        uint32_t height);
+typedef void (*LOCALDATACHANNELREADY_CALLBACK)();
+typedef void (*DATAFROMEDATECHANNELREADY_CALLBACK)(const char* msg);
+typedef void (*FAILURE_CALLBACK)(const char* msg);
+typedef void (*LOCALSDPREADYTOSEND_CALLBACK)(const char* type, const char* sdp);
+typedef void (*ICECANDIDATEREADYTOSEND_CALLBACK)(const char* candidate,
+                                                 int sdp_mline_index,
+                                                 const char* sdp_mid);
+typedef void (*AUDIOBUSREADY_CALLBACK)(const void* audio_data,
+                                       int bits_per_sample,
+                                       int sample_rate,
+                                       int number_of_channels,
+                                       int number_of_frames);
+
+#if defined(WEBRTC_WIN)
+#define WEBRTC_PLUGIN_API __declspec(dllexport)
+#elif defined(WEBRTC_ANDROID)
+#define WEBRTC_PLUGIN_API __attribute__((visibility("default")))
+#endif
+extern "C" {
+// Create a peerconnection and return a unique peer connection id.
+WEBRTC_PLUGIN_API int CreatePeerConnection(const char** turn_urls,
+                                           int no_of_urls,
+                                           const char* username,
+                                           const char* credential,
+                                           bool mandatory_receive_video);
+// Close a peerconnection.
+WEBRTC_PLUGIN_API bool ClosePeerConnection(int peer_connection_id);
+// Add a audio stream. If audio_only is true, the stream only has an audio
+// track and no video track.
+WEBRTC_PLUGIN_API bool AddStream(int peer_connection_id, bool audio_only);
+// Add a data channel to peer connection.
+WEBRTC_PLUGIN_API bool AddDataChannel(int peer_connection_id);
+// Create a peer connection offer.
+WEBRTC_PLUGIN_API bool CreateOffer(int peer_connection_id);
+// Create a peer connection answer.
+WEBRTC_PLUGIN_API bool CreateAnswer(int peer_connection_id);
+// Send data through data channel.
+WEBRTC_PLUGIN_API bool SendDataViaDataChannel(int peer_connection_id,
+                                              const char* data);
+// Set audio control. If is_mute=true, no audio will playout. If is_record=true,
+// AUDIOBUSREADY_CALLBACK will be called every 10 ms.
+WEBRTC_PLUGIN_API bool SetAudioControl(int peer_connection_id,
+                                       bool is_mute,
+                                       bool is_record);
+// Set remote sdp.
+WEBRTC_PLUGIN_API bool SetRemoteDescription(int peer_connection_id,
+                                            const char* type,
+                                            const char* sdp);
+// Add ice candidate.
+WEBRTC_PLUGIN_API bool AddIceCandidate(int peer_connection_id,
+                                       const char* candidate,
+                                       int sdp_mlineindex,
+                                       const char* sdp_mid);
+
+// Register callback functions.
+WEBRTC_PLUGIN_API bool RegisterOnLocalI420FrameReady(
+    int peer_connection_id,
+    I420FRAMEREADY_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnRemoteI420FrameReady(
+    int peer_connection_id,
+    I420FRAMEREADY_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnLocalDataChannelReady(
+    int peer_connection_id,
+    LOCALDATACHANNELREADY_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnDataFromDataChannelReady(
+    int peer_connection_id,
+    DATAFROMEDATECHANNELREADY_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnFailure(int peer_connection_id,
+                                         FAILURE_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnAudioBusReady(int peer_connection_id,
+                                               AUDIOBUSREADY_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnLocalSdpReadytoSend(
+    int peer_connection_id,
+    LOCALSDPREADYTOSEND_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnIceCandidateReadytoSend(
+    int peer_connection_id,
+    ICECANDIDATEREADYTOSEND_CALLBACK callback);
+}
+
+#endif  // EXAMPLES_UNITYPLUGIN_UNITY_PLUGIN_APIS_H_
diff --git a/third_party/libwebrtc/examples/unityplugin/video_observer.cc b/third_party/libwebrtc/examples/unityplugin/video_observer.cc
new file mode 100644
index 0000000000..7e33b08e27
--- /dev/null
+++ b/third_party/libwebrtc/examples/unityplugin/video_observer.cc
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "examples/unityplugin/video_observer.h"
+
+void VideoObserver::SetVideoCallback(I420FRAMEREADY_CALLBACK callback) {
+  std::lock_guard<std::mutex> lock(mutex);
+  OnI420FrameReady = callback;
+}
+
+void VideoObserver::OnFrame(const webrtc::VideoFrame& frame) {
+  std::unique_lock<std::mutex> lock(mutex);
+  if (!OnI420FrameReady)
+    return;
+
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer(
+      frame.video_frame_buffer());
+
+  if (buffer->type() != webrtc::VideoFrameBuffer::Type::kI420A) {
+    rtc::scoped_refptr<webrtc::I420BufferInterface> i420_buffer =
+        buffer->ToI420();
+    OnI420FrameReady(i420_buffer->DataY(), i420_buffer->DataU(),
+                     i420_buffer->DataV(), nullptr, i420_buffer->StrideY(),
+                     i420_buffer->StrideU(), i420_buffer->StrideV(), 0,
+                     frame.width(), frame.height());
+
+  } else {
+    // The buffer has alpha channel.
+    const webrtc::I420ABufferInterface* i420a_buffer = buffer->GetI420A();
+
+    OnI420FrameReady(i420a_buffer->DataY(), i420a_buffer->DataU(),
+                     i420a_buffer->DataV(), i420a_buffer->DataA(),
+                     i420a_buffer->StrideY(), i420a_buffer->StrideU(),
+                     i420a_buffer->StrideV(), i420a_buffer->StrideA(),
+                     frame.width(), frame.height());
+  }
+}
diff --git a/third_party/libwebrtc/examples/unityplugin/video_observer.h b/third_party/libwebrtc/examples/unityplugin/video_observer.h
new file mode 100644
index 0000000000..01ccd2191a
--- /dev/null
+++ b/third_party/libwebrtc/examples/unityplugin/video_observer.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef EXAMPLES_UNITYPLUGIN_VIDEO_OBSERVER_H_
+#define EXAMPLES_UNITYPLUGIN_VIDEO_OBSERVER_H_
+
+#include <mutex>
+
+#include "api/media_stream_interface.h"
+#include "api/video/video_sink_interface.h"
+#include "examples/unityplugin/unity_plugin_apis.h"
+
+class VideoObserver : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+ public:
+  VideoObserver() {}
+  ~VideoObserver() {}
+  void SetVideoCallback(I420FRAMEREADY_CALLBACK callback);
+
+ protected:
+  // VideoSinkInterface implementation
+  void OnFrame(const webrtc::VideoFrame& frame) override;
+
+ private:
+  I420FRAMEREADY_CALLBACK OnI420FrameReady = nullptr;
+  std::mutex mutex;
+};
+
+#endif  // EXAMPLES_UNITYPLUGIN_VIDEO_OBSERVER_H_