Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

1 files changed, 1167 insertions, 0 deletions

diff --git a/third_party/libwebrtc/pc/data_channel_integrationtest.cc b/third_party/libwebrtc/pc/data_channel_integrationtest.cc
new file mode 100644
index 0000000000..faec76d03e
--- /dev/null
+++ b/third_party/libwebrtc/pc/data_channel_integrationtest.cc
@@ -0,0 +1,1167 @@
+/*
+ *  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 <stdint.h>
+
+#include <cstdlib>
+#include <iterator>
+#include <string>
+#include <tuple>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/types/optional.h"
+#include "api/data_channel_interface.h"
+#include "api/dtls_transport_interface.h"
+#include "api/peer_connection_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sctp_transport_interface.h"
+#include "api/stats/rtc_stats_report.h"
+#include "api/stats/rtcstats_objects.h"
+#include "api/units/time_delta.h"
+#include "p2p/base/transport_description.h"
+#include "p2p/base/transport_info.h"
+#include "pc/media_session.h"
+#include "pc/session_description.h"
+#include "pc/test/integration_test_helpers.h"
+#include "pc/test/mock_peer_connection_observers.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/fake_clock.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/helpers.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/virtual_socket_server.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+// All tests in this file require SCTP support.
+#ifdef WEBRTC_HAVE_SCTP
+
+#if defined(WEBRTC_ANDROID)
+// Disable heavy tests running on low-end Android devices.
+#define DISABLED_ON_ANDROID(t) DISABLED_##t
+#else
+#define DISABLED_ON_ANDROID(t) t
+#endif
+
+class DataChannelIntegrationTest
+    : public PeerConnectionIntegrationBaseTest,
+      public ::testing::WithParamInterface<std::tuple<SdpSemantics, bool>> {
+ protected:
+  DataChannelIntegrationTest()
+      : PeerConnectionIntegrationBaseTest(std::get<0>(GetParam())),
+        allow_media_(std::get<1>(GetParam())) {}
+  bool allow_media() { return allow_media_; }
+
+  bool CreatePeerConnectionWrappers() {
+    if (allow_media_) {
+      return PeerConnectionIntegrationBaseTest::CreatePeerConnectionWrappers();
+    }
+    return PeerConnectionIntegrationBaseTest::
+        CreatePeerConnectionWrappersWithoutMediaEngine();
+  }
+
+ private:
+  // True if media is allowed to be added
+  const bool allow_media_;
+};
+
+// Fake clock must be set before threads are started to prevent race on
+// Set/GetClockForTesting().
+// To achieve that, multiple inheritance is used as a mixin pattern
+// where order of construction is finely controlled.
+// This also ensures peerconnection is closed before switching back to non-fake
+// clock, avoiding other races and DCHECK failures such as in rtp_sender.cc.
+class FakeClockForTest : public rtc::ScopedFakeClock {
+ protected:
+  FakeClockForTest() {
+    // Some things use a time of "0" as a special value, so we need to start out
+    // the fake clock at a nonzero time.
+    // TODO(deadbeef): Fix this.
+    AdvanceTime(webrtc::TimeDelta::Seconds(1));
+  }
+
+  // Explicit handle.
+  ScopedFakeClock& FakeClock() { return *this; }
+};
+
+class DataChannelIntegrationTestPlanB
+    : public PeerConnectionIntegrationBaseTest {
+ protected:
+  DataChannelIntegrationTestPlanB()
+      : PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB_DEPRECATED) {}
+};
+
+class DataChannelIntegrationTestUnifiedPlan
+    : public PeerConnectionIntegrationBaseTest {
+ protected:
+  DataChannelIntegrationTestUnifiedPlan()
+      : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {}
+};
+
+void MakeActiveSctpOffer(cricket::SessionDescription* desc) {
+  auto& transport_infos = desc->transport_infos();
+  for (auto& transport_info : transport_infos) {
+    transport_info.description.connection_role = cricket::CONNECTIONROLE_ACTIVE;
+  }
+}
+
+// This test causes a PeerConnection to enter Disconnected state, and
+// sends data on a DataChannel while disconnected.
+// The data should be surfaced when the connection reestablishes.
+TEST_P(DataChannelIntegrationTest, DataChannelWhileDisconnected) {
+  CreatePeerConnectionWrappers();
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer(), kDefaultTimeout);
+  std::string data1 = "hello first";
+  caller()->data_channel()->Send(DataBuffer(data1));
+  EXPECT_EQ_WAIT(data1, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  // Cause a network outage
+  virtual_socket_server()->set_drop_probability(1.0);
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
+                 caller()->standardized_ice_connection_state(),
+                 kDefaultTimeout);
+  std::string data2 = "hello second";
+  caller()->data_channel()->Send(DataBuffer(data2));
+  // Remove the network outage. The connection should reestablish.
+  virtual_socket_server()->set_drop_probability(0.0);
+  EXPECT_EQ_WAIT(data2, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+}
+
+// This test causes a PeerConnection to enter Disconnected state,
+// sends data on a DataChannel while disconnected, and then triggers
+// an ICE restart.
+// The data should be surfaced when the connection reestablishes.
+TEST_P(DataChannelIntegrationTest, DataChannelWhileDisconnectedIceRestart) {
+  CreatePeerConnectionWrappers();
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer(), kDefaultTimeout);
+  std::string data1 = "hello first";
+  caller()->data_channel()->Send(DataBuffer(data1));
+  EXPECT_EQ_WAIT(data1, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  // Cause a network outage
+  virtual_socket_server()->set_drop_probability(1.0);
+  ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
+                 caller()->standardized_ice_connection_state(),
+                 kDefaultTimeout);
+  std::string data2 = "hello second";
+  caller()->data_channel()->Send(DataBuffer(data2));
+
+  // Trigger an ICE restart. The signaling channel is not affected by
+  // the network outage.
+  caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Remove the network outage. The connection should reestablish.
+  virtual_socket_server()->set_drop_probability(0.0);
+  EXPECT_EQ_WAIT(data2, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+}
+
+// This test sets up a call between two parties with audio, video and an SCTP
+// data channel.
+TEST_P(DataChannelIntegrationTest, EndToEndCallWithSctpDataChannel) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  // Expect that data channel created on caller side will show up for callee as
+  // well.
+  caller()->CreateDataChannel();
+  if (allow_media()) {
+    caller()->AddAudioVideoTracks();
+    callee()->AddAudioVideoTracks();
+  }
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  if (allow_media()) {
+    // Ensure the existence of the SCTP data channel didn't impede audio/video.
+    MediaExpectations media_expectations;
+    media_expectations.ExpectBidirectionalAudioAndVideo();
+    ASSERT_TRUE(ExpectNewFrames(media_expectations));
+  }
+  // Caller data channel should already exist (it created one). Callee data
+  // channel may not exist yet, since negotiation happens in-band, not in SDP.
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  // Ensure data can be sent in both directions.
+  std::string data = "hello world";
+  caller()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  callee()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
+                 kDefaultTimeout);
+}
+
+// This test sets up a call between two parties with an SCTP
+// data channel only, and sends messages of various sizes.
+TEST_P(DataChannelIntegrationTest,
+       EndToEndCallWithSctpDataChannelVariousSizes) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  // Expect that data channel created on caller side will show up for callee as
+  // well.
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Caller data channel should already exist (it created one). Callee data
+  // channel may not exist yet, since negotiation happens in-band, not in SDP.
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  for (int message_size = 1; message_size < 100000; message_size *= 2) {
+    std::string data(message_size, 'a');
+    caller()->data_channel()->Send(DataBuffer(data));
+    EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
+                   kDefaultTimeout);
+    callee()->data_channel()->Send(DataBuffer(data));
+    EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
+                   kDefaultTimeout);
+  }
+  // Specifically probe the area around the MTU size.
+  for (int message_size = 1100; message_size < 1300; message_size += 1) {
+    std::string data(message_size, 'a');
+    caller()->data_channel()->Send(DataBuffer(data));
+    EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
+                   kDefaultTimeout);
+    callee()->data_channel()->Send(DataBuffer(data));
+    EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
+                   kDefaultTimeout);
+  }
+}
+
+// This test sets up a call between two parties with an SCTP
+// data channel only, and sends empty messages
+TEST_P(DataChannelIntegrationTest,
+       EndToEndCallWithSctpDataChannelEmptyMessages) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  // Expect that data channel created on caller side will show up for callee as
+  // well.
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Caller data channel should already exist (it created one). Callee data
+  // channel may not exist yet, since negotiation happens in-band, not in SDP.
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  // Ensure data can be sent in both directions.
+  // Sending empty string data
+  std::string data = "";
+  caller()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(1u, callee()->data_observer()->received_message_count(),
+                 kDefaultTimeout);
+  EXPECT_TRUE(callee()->data_observer()->last_message().empty());
+  EXPECT_FALSE(callee()->data_observer()->messages().back().binary);
+  callee()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(1u, caller()->data_observer()->received_message_count(),
+                 kDefaultTimeout);
+  EXPECT_TRUE(caller()->data_observer()->last_message().empty());
+  EXPECT_FALSE(caller()->data_observer()->messages().back().binary);
+
+  // Sending empty binary data
+  rtc::CopyOnWriteBuffer empty_buffer;
+  caller()->data_channel()->Send(DataBuffer(empty_buffer, true));
+  EXPECT_EQ_WAIT(2u, callee()->data_observer()->received_message_count(),
+                 kDefaultTimeout);
+  EXPECT_TRUE(callee()->data_observer()->last_message().empty());
+  EXPECT_TRUE(callee()->data_observer()->messages().back().binary);
+  callee()->data_channel()->Send(DataBuffer(empty_buffer, true));
+  EXPECT_EQ_WAIT(2u, caller()->data_observer()->received_message_count(),
+                 kDefaultTimeout);
+  EXPECT_TRUE(caller()->data_observer()->last_message().empty());
+  EXPECT_TRUE(caller()->data_observer()->messages().back().binary);
+}
+
+TEST_P(DataChannelIntegrationTest,
+       EndToEndCallWithSctpDataChannelLowestSafeMtu) {
+  // The lowest payload size limit that's tested and found safe for this
+  // application. Note that this is not the safe limit under all conditions;
+  // in particular, the default is not the largest DTLS signature, and
+  // this test does not use TURN.
+  const size_t kLowestSafePayloadSizeLimit = 1225;
+
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  // Expect that data channel created on caller side will show up for callee as
+  // well.
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Caller data channel should already exist (it created one). Callee data
+  // channel may not exist yet, since negotiation happens in-band, not in SDP.
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  virtual_socket_server()->set_max_udp_payload(kLowestSafePayloadSizeLimit);
+  for (int message_size = 1140; message_size < 1240; message_size += 1) {
+    std::string data(message_size, 'a');
+    caller()->data_channel()->Send(DataBuffer(data));
+    ASSERT_EQ_WAIT(data, callee()->data_observer()->last_message(),
+                   kDefaultTimeout);
+    callee()->data_channel()->Send(DataBuffer(data));
+    ASSERT_EQ_WAIT(data, caller()->data_observer()->last_message(),
+                   kDefaultTimeout);
+  }
+}
+
+// This test verifies that lowering the MTU of the connection will cause
+// the datachannel to not transmit reliably.
+// The purpose of this test is to ensure that we know how a too-small MTU
+// error manifests itself.
+TEST_P(DataChannelIntegrationTest, EndToEndCallWithSctpDataChannelHarmfulMtu) {
+  // The lowest payload size limit that's tested and found safe for this
+  // application in this configuration (see test above).
+  const size_t kLowestSafePayloadSizeLimit = 1225;
+  // The size of the smallest message that fails to be delivered.
+  const size_t kMessageSizeThatIsNotDelivered = 1157;
+
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  virtual_socket_server()->set_max_udp_payload(kLowestSafePayloadSizeLimit - 1);
+  // Probe for an undelivered or slowly delivered message. The exact
+  // size limit seems to be dependent on the message history, so make the
+  // code easily able to find the current value.
+  bool failure_seen = false;
+  for (size_t message_size = 1110; message_size < 1400; message_size++) {
+    const size_t message_count =
+        callee()->data_observer()->received_message_count();
+    const std::string data(message_size, 'a');
+    caller()->data_channel()->Send(DataBuffer(data));
+    // Wait a very short time for the message to be delivered.
+    // Note: Waiting only 10 ms is too short for Windows bots; they will
+    // flakily fail at a random frame.
+    WAIT(callee()->data_observer()->received_message_count() > message_count,
+         100);
+    if (callee()->data_observer()->received_message_count() == message_count) {
+      ASSERT_EQ(kMessageSizeThatIsNotDelivered, message_size);
+      failure_seen = true;
+      break;
+    }
+  }
+  ASSERT_TRUE(failure_seen);
+}
+
+// Ensure that when the callee closes an SCTP data channel, the closing
+// procedure results in the data channel being closed for the caller as well.
+TEST_P(DataChannelIntegrationTest, CalleeClosesSctpDataChannel) {
+  // Same procedure as above test.
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  if (allow_media()) {
+    caller()->AddAudioVideoTracks();
+    callee()->AddAudioVideoTracks();
+  }
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  // Close the data channel on the callee side, and wait for it to reach the
+  // "closed" state on both sides.
+  callee()->data_channel()->Close();
+
+  DataChannelInterface::DataState expected_states[] = {
+      DataChannelInterface::DataState::kConnecting,
+      DataChannelInterface::DataState::kOpen,
+      DataChannelInterface::DataState::kClosing,
+      DataChannelInterface::DataState::kClosed};
+
+  EXPECT_EQ_WAIT(DataChannelInterface::DataState::kClosed,
+                 caller()->data_observer()->state(), kDefaultTimeout);
+  EXPECT_THAT(caller()->data_observer()->states(),
+              ::testing::ElementsAreArray(expected_states));
+
+  EXPECT_EQ_WAIT(DataChannelInterface::DataState::kClosed,
+                 callee()->data_observer()->state(), kDefaultTimeout);
+  EXPECT_THAT(callee()->data_observer()->states(),
+              ::testing::ElementsAreArray(expected_states));
+}
+
+TEST_P(DataChannelIntegrationTest, SctpDataChannelConfigSentToOtherSide) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  webrtc::DataChannelInit init;
+  init.id = 53;
+  init.maxRetransmits = 52;
+  caller()->CreateDataChannel("data-channel", &init);
+  if (allow_media()) {
+    caller()->AddAudioVideoTracks();
+    callee()->AddAudioVideoTracks();
+  }
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+  // Since "negotiated" is false, the "id" parameter should be ignored.
+  EXPECT_NE(init.id, callee()->data_channel()->id());
+  EXPECT_EQ("data-channel", callee()->data_channel()->label());
+  EXPECT_EQ(init.maxRetransmits, callee()->data_channel()->maxRetransmits());
+  EXPECT_FALSE(callee()->data_channel()->negotiated());
+}
+
+// Test sctp's ability to process unordered data stream, where data actually
+// arrives out of order using simulated delays. Previously there have been some
+// bugs in this area.
+TEST_P(DataChannelIntegrationTest, StressTestUnorderedSctpDataChannel) {
+  // Introduce random network delays.
+  // Otherwise it's not a true "unordered" test.
+  virtual_socket_server()->set_delay_mean(20);
+  virtual_socket_server()->set_delay_stddev(5);
+  virtual_socket_server()->UpdateDelayDistribution();
+  // Normal procedure, but with unordered data channel config.
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  webrtc::DataChannelInit init;
+  init.ordered = false;
+  caller()->CreateDataChannel(&init);
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  static constexpr int kNumMessages = 100;
+  // Deliberately chosen to be larger than the MTU so messages get fragmented.
+  static constexpr size_t kMaxMessageSize = 4096;
+  // Create and send random messages.
+  std::vector<std::string> sent_messages;
+  for (int i = 0; i < kNumMessages; ++i) {
+    size_t length =
+        (rand() % kMaxMessageSize) + 1;  // NOLINT (rand_r instead of rand)
+    std::string message;
+    ASSERT_TRUE(rtc::CreateRandomString(length, &message));
+    caller()->data_channel()->Send(DataBuffer(message));
+    callee()->data_channel()->Send(DataBuffer(message));
+    sent_messages.push_back(message);
+  }
+
+  // Wait for all messages to be received.
+  EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
+                 caller()->data_observer()->received_message_count(),
+                 kDefaultTimeout);
+  EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
+                 callee()->data_observer()->received_message_count(),
+                 kDefaultTimeout);
+
+  // Sort and compare to make sure none of the messages were corrupted.
+  std::vector<std::string> caller_received_messages;
+  absl::c_transform(caller()->data_observer()->messages(),
+                    std::back_inserter(caller_received_messages),
+                    [](const auto& a) { return a.data; });
+
+  std::vector<std::string> callee_received_messages;
+  absl::c_transform(callee()->data_observer()->messages(),
+                    std::back_inserter(callee_received_messages),
+                    [](const auto& a) { return a.data; });
+
+  absl::c_sort(sent_messages);
+  absl::c_sort(caller_received_messages);
+  absl::c_sort(callee_received_messages);
+  EXPECT_EQ(sent_messages, caller_received_messages);
+  EXPECT_EQ(sent_messages, callee_received_messages);
+}
+
+// Repeatedly open and close data channels on a peer connection to check that
+// the channels are properly negotiated and SCTP stream IDs properly recycled.
+TEST_P(DataChannelIntegrationTest, StressTestOpenCloseChannelNoDelay) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+
+  int channel_id = 0;
+  const size_t kChannelCount = 8;
+  const size_t kIterations = 10;
+  bool has_negotiated = false;
+
+  webrtc::DataChannelInit init;
+  for (size_t repeats = 0; repeats < kIterations; ++repeats) {
+    RTC_LOG(LS_INFO) << "Iteration " << (repeats + 1) << "/" << kIterations;
+
+    for (size_t i = 0; i < kChannelCount; ++i) {
+      rtc::StringBuilder sb;
+      sb << "channel-" << channel_id++;
+      caller()->CreateDataChannel(sb.Release(), &init);
+    }
+    ASSERT_EQ(caller()->data_channels().size(), kChannelCount);
+
+    if (!has_negotiated) {
+      caller()->CreateAndSetAndSignalOffer();
+      ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+      has_negotiated = true;
+    }
+
+    for (size_t i = 0; i < kChannelCount; ++i) {
+      ASSERT_EQ_WAIT(caller()->data_channels()[i]->state(),
+                     DataChannelInterface::DataState::kOpen, kDefaultTimeout);
+      RTC_LOG(LS_INFO) << "Caller Channel "
+                       << caller()->data_channels()[i]->label() << " with id "
+                       << caller()->data_channels()[i]->id() << " is open.";
+    }
+    ASSERT_EQ_WAIT(callee()->data_channels().size(), kChannelCount,
+                   kDefaultTimeout);
+    for (size_t i = 0; i < kChannelCount; ++i) {
+      ASSERT_EQ_WAIT(callee()->data_channels()[i]->state(),
+                     DataChannelInterface::DataState::kOpen, kDefaultTimeout);
+      RTC_LOG(LS_INFO) << "Callee Channel "
+                       << callee()->data_channels()[i]->label() << " with id "
+                       << callee()->data_channels()[i]->id() << " is open.";
+    }
+
+    // Closing from both sides to attempt creating races.
+    // A real application would likely only close from one side.
+    for (size_t i = 0; i < kChannelCount; ++i) {
+      if (i % 3 == 0) {
+        callee()->data_channels()[i]->Close();
+        caller()->data_channels()[i]->Close();
+      } else {
+        caller()->data_channels()[i]->Close();
+        callee()->data_channels()[i]->Close();
+      }
+    }
+
+    for (size_t i = 0; i < kChannelCount; ++i) {
+      ASSERT_EQ_WAIT(caller()->data_channels()[i]->state(),
+                     DataChannelInterface::DataState::kClosed, kDefaultTimeout);
+      ASSERT_EQ_WAIT(callee()->data_channels()[i]->state(),
+                     DataChannelInterface::DataState::kClosed, kDefaultTimeout);
+    }
+
+    caller()->data_channels().clear();
+    caller()->data_observers().clear();
+    callee()->data_channels().clear();
+    callee()->data_observers().clear();
+  }
+}
+
+// Repeatedly open and close data channels on a peer connection to check that
+// the channels are properly negotiated and SCTP stream IDs properly recycled.
+// Some delay is added for better coverage.
+TEST_P(DataChannelIntegrationTest, StressTestOpenCloseChannelWithDelay) {
+  // Simulate some network delay
+  virtual_socket_server()->set_delay_mean(20);
+  virtual_socket_server()->set_delay_stddev(5);
+  virtual_socket_server()->UpdateDelayDistribution();
+
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+
+  int channel_id = 0;
+  const size_t kChannelCount = 8;
+  const size_t kIterations = 10;
+  bool has_negotiated = false;
+
+  webrtc::DataChannelInit init;
+  for (size_t repeats = 0; repeats < kIterations; ++repeats) {
+    RTC_LOG(LS_INFO) << "Iteration " << (repeats + 1) << "/" << kIterations;
+
+    for (size_t i = 0; i < kChannelCount; ++i) {
+      rtc::StringBuilder sb;
+      sb << "channel-" << channel_id++;
+      caller()->CreateDataChannel(sb.Release(), &init);
+    }
+    ASSERT_EQ(caller()->data_channels().size(), kChannelCount);
+
+    if (!has_negotiated) {
+      caller()->CreateAndSetAndSignalOffer();
+      ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+      has_negotiated = true;
+    }
+
+    for (size_t i = 0; i < kChannelCount; ++i) {
+      ASSERT_EQ_WAIT(caller()->data_channels()[i]->state(),
+                     DataChannelInterface::DataState::kOpen, kDefaultTimeout);
+      RTC_LOG(LS_INFO) << "Caller Channel "
+                       << caller()->data_channels()[i]->label() << " with id "
+                       << caller()->data_channels()[i]->id() << " is open.";
+    }
+    ASSERT_EQ_WAIT(callee()->data_channels().size(), kChannelCount,
+                   kDefaultTimeout);
+    for (size_t i = 0; i < kChannelCount; ++i) {
+      ASSERT_EQ_WAIT(callee()->data_channels()[i]->state(),
+                     DataChannelInterface::DataState::kOpen, kDefaultTimeout);
+      RTC_LOG(LS_INFO) << "Callee Channel "
+                       << callee()->data_channels()[i]->label() << " with id "
+                       << callee()->data_channels()[i]->id() << " is open.";
+    }
+
+    // Closing from both sides to attempt creating races.
+    // A real application would likely only close from one side.
+    for (size_t i = 0; i < kChannelCount; ++i) {
+      if (i % 3 == 0) {
+        callee()->data_channels()[i]->Close();
+        caller()->data_channels()[i]->Close();
+      } else {
+        caller()->data_channels()[i]->Close();
+        callee()->data_channels()[i]->Close();
+      }
+    }
+
+    for (size_t i = 0; i < kChannelCount; ++i) {
+      ASSERT_EQ_WAIT(caller()->data_channels()[i]->state(),
+                     DataChannelInterface::DataState::kClosed, kDefaultTimeout);
+      ASSERT_EQ_WAIT(callee()->data_channels()[i]->state(),
+                     DataChannelInterface::DataState::kClosed, kDefaultTimeout);
+    }
+
+    caller()->data_channels().clear();
+    caller()->data_observers().clear();
+    callee()->data_channels().clear();
+    callee()->data_observers().clear();
+  }
+}
+
+// This test sets up a call between two parties with audio, and video. When
+// audio and video are setup and flowing, an SCTP data channel is negotiated.
+TEST_P(DataChannelIntegrationTest, AddSctpDataChannelInSubsequentOffer) {
+  // This test can't be performed without media.
+  if (!allow_media()) {
+    return;
+  }
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  // Do initial offer/answer with audio/video.
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Create data channel and do new offer and answer.
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Caller data channel should already exist (it created one). Callee data
+  // channel may not exist yet, since negotiation happens in-band, not in SDP.
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+  // Ensure data can be sent in both directions.
+  std::string data = "hello world";
+  caller()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  callee()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
+                 kDefaultTimeout);
+}
+
+// Set up a connection initially just using SCTP data channels, later
+// upgrading to audio/video, ensuring frames are received end-to-end.
+// Effectively the inverse of the test above. This was broken in M57; see
+// https://crbug.com/711243
+TEST_P(DataChannelIntegrationTest, SctpDataChannelToAudioVideoUpgrade) {
+  // This test can't be performed without media.
+  if (!allow_media()) {
+    return;
+  }
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  // Do initial offer/answer with just data channel.
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Wait until data can be sent over the data channel.
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  // Do subsequent offer/answer with two-way audio and video. Audio and video
+  // should end up bundled on the DTLS/ICE transport already used for data.
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  MediaExpectations media_expectations;
+  media_expectations.ExpectBidirectionalAudioAndVideo();
+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
+}
+
+static void MakeSpecCompliantSctpOffer(cricket::SessionDescription* desc) {
+  cricket::SctpDataContentDescription* dcd_offer =
+      GetFirstSctpDataContentDescription(desc);
+  // See https://crbug.com/webrtc/11211 - this function is a no-op
+  ASSERT_TRUE(dcd_offer);
+  dcd_offer->set_use_sctpmap(false);
+  dcd_offer->set_protocol("UDP/DTLS/SCTP");
+}
+
+// Test that the data channel works when a spec-compliant SCTP m= section is
+// offered (using "a=sctp-port" instead of "a=sctpmap", and using
+// "UDP/DTLS/SCTP" as the protocol).
+TEST_P(DataChannelIntegrationTest,
+       DataChannelWorksWhenSpecCompliantSctpOfferReceived) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->SetGeneratedSdpMunger(MakeSpecCompliantSctpOffer);
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  // Ensure data can be sent in both directions.
+  std::string data = "hello world";
+  caller()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  callee()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
+                 kDefaultTimeout);
+}
+
+// Test that after closing PeerConnections, they stop sending any packets
+// (ICE, DTLS, RTP...).
+TEST_P(DataChannelIntegrationTest, ClosingConnectionStopsPacketFlow) {
+  // This test can't be performed without media.
+  if (!allow_media()) {
+    return;
+  }
+  // Set up audio/video/data, wait for some frames to be received.
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->AddAudioVideoTracks();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  MediaExpectations media_expectations;
+  media_expectations.CalleeExpectsSomeAudioAndVideo();
+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
+  // Close PeerConnections.
+  ClosePeerConnections();
+  // Pump messages for a second, and ensure no new packets end up sent.
+  uint32_t sent_packets_a = virtual_socket_server()->sent_packets();
+  WAIT(false, 1000);
+  uint32_t sent_packets_b = virtual_socket_server()->sent_packets();
+  EXPECT_EQ(sent_packets_a, sent_packets_b);
+}
+
+TEST_P(DataChannelIntegrationTest, DtlsRoleIsSetNormally) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  ASSERT_FALSE(caller()->pc()->GetSctpTransport());
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  ASSERT_TRUE(caller()->pc()->GetSctpTransport());
+  ASSERT_TRUE(
+      caller()->pc()->GetSctpTransport()->Information().dtls_transport());
+  EXPECT_TRUE(caller()
+                  ->pc()
+                  ->GetSctpTransport()
+                  ->Information()
+                  .dtls_transport()
+                  ->Information()
+                  .role());
+  EXPECT_EQ(caller()
+                ->pc()
+                ->GetSctpTransport()
+                ->Information()
+                .dtls_transport()
+                ->Information()
+                .role(),
+            DtlsTransportTlsRole::kServer);
+  EXPECT_EQ(callee()
+                ->pc()
+                ->GetSctpTransport()
+                ->Information()
+                .dtls_transport()
+                ->Information()
+                .role(),
+            DtlsTransportTlsRole::kClient);
+  // ID should be assigned according to the odd/even rule based on role;
+  // client gets even numbers, server gets odd ones. RFC 8832 section 6.
+  // TODO(hta): Test multiple channels.
+  EXPECT_EQ(caller()->data_channel()->id(), 1);
+}
+
+TEST_P(DataChannelIntegrationTest, DtlsRoleIsSetWhenReversed) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  callee()->SetReceivedSdpMunger(MakeActiveSctpOffer);
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE(caller()
+                  ->pc()
+                  ->GetSctpTransport()
+                  ->Information()
+                  .dtls_transport()
+                  ->Information()
+                  .role());
+  EXPECT_EQ(caller()
+                ->pc()
+                ->GetSctpTransport()
+                ->Information()
+                .dtls_transport()
+                ->Information()
+                .role(),
+            DtlsTransportTlsRole::kClient);
+  EXPECT_EQ(callee()
+                ->pc()
+                ->GetSctpTransport()
+                ->Information()
+                .dtls_transport()
+                ->Information()
+                .role(),
+            DtlsTransportTlsRole::kServer);
+  // ID should be assigned according to the odd/even rule based on role;
+  // client gets even numbers, server gets odd ones. RFC 8832 section 6.
+  // TODO(hta): Test multiple channels.
+  EXPECT_EQ(caller()->data_channel()->id(), 0);
+}
+
+TEST_P(DataChannelIntegrationTest,
+       DtlsRoleIsSetWhenReversedWithChannelCollision) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+
+  callee()->SetReceivedSdpMunger([this](cricket::SessionDescription* desc) {
+    MakeActiveSctpOffer(desc);
+    callee()->CreateDataChannel();
+  });
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  ASSERT_EQ_WAIT(callee()->data_channels().size(), 2U, kDefaultTimeout);
+  ASSERT_EQ_WAIT(caller()->data_channels().size(), 2U, kDefaultTimeout);
+  EXPECT_TRUE(caller()
+                  ->pc()
+                  ->GetSctpTransport()
+                  ->Information()
+                  .dtls_transport()
+                  ->Information()
+                  .role());
+  EXPECT_EQ(caller()
+                ->pc()
+                ->GetSctpTransport()
+                ->Information()
+                .dtls_transport()
+                ->Information()
+                .role(),
+            DtlsTransportTlsRole::kClient);
+  EXPECT_EQ(callee()
+                ->pc()
+                ->GetSctpTransport()
+                ->Information()
+                .dtls_transport()
+                ->Information()
+                .role(),
+            DtlsTransportTlsRole::kServer);
+  // ID should be assigned according to the odd/even rule based on role;
+  // client gets even numbers, server gets odd ones. RFC 8832 section 6.
+  ASSERT_EQ(caller()->data_channels().size(), 2U);
+  ASSERT_EQ(callee()->data_channels().size(), 2U);
+  EXPECT_EQ(caller()->data_channels()[0]->id(), 0);
+  EXPECT_EQ(caller()->data_channels()[1]->id(), 1);
+  EXPECT_EQ(callee()->data_channels()[0]->id(), 1);
+  EXPECT_EQ(callee()->data_channels()[1]->id(), 0);
+}
+
+// Test that transport stats are generated by the RTCStatsCollector for a
+// connection that only involves data channels. This is a regression test for
+// crbug.com/826972.
+TEST_P(DataChannelIntegrationTest,
+       TransportStatsReportedForDataChannelOnlyConnection) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
+
+  auto caller_report = caller()->NewGetStats();
+  EXPECT_EQ(1u, caller_report->GetStatsOfType<RTCTransportStats>().size());
+  auto callee_report = callee()->NewGetStats();
+  EXPECT_EQ(1u, callee_report->GetStatsOfType<RTCTransportStats>().size());
+}
+
+TEST_P(DataChannelIntegrationTest, QueuedPacketsGetDeliveredInReliableMode) {
+  CreatePeerConnectionWrappers();
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
+
+  caller()->data_channel()->Send(DataBuffer("hello first"));
+  ASSERT_EQ_WAIT(1u, callee()->data_observer()->received_message_count(),
+                 kDefaultTimeout);
+  // Cause a temporary network outage
+  virtual_socket_server()->set_drop_probability(1.0);
+  for (int i = 1; i <= 10; i++) {
+    caller()->data_channel()->Send(DataBuffer("Sent while blocked"));
+  }
+  // Nothing should be delivered during outage. Short wait.
+  EXPECT_EQ_WAIT(1u, callee()->data_observer()->received_message_count(), 10);
+  // Reverse outage
+  virtual_socket_server()->set_drop_probability(0.0);
+  // All packets should be delivered.
+  EXPECT_EQ_WAIT(11u, callee()->data_observer()->received_message_count(),
+                 kDefaultTimeout);
+}
+
+TEST_P(DataChannelIntegrationTest, QueuedPacketsGetDroppedInUnreliableMode) {
+  CreatePeerConnectionWrappers();
+  ConnectFakeSignaling();
+  DataChannelInit init;
+  init.maxRetransmits = 0;
+  init.ordered = false;
+  caller()->CreateDataChannel(&init);
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
+  caller()->data_channel()->Send(DataBuffer("hello first"));
+  ASSERT_EQ_WAIT(1u, callee()->data_observer()->received_message_count(),
+                 kDefaultTimeout);
+  // Cause a temporary network outage
+  virtual_socket_server()->set_drop_probability(1.0);
+  // Send a few packets. Note that all get dropped only when all packets
+  // fit into the receiver receive window/congestion window, so that they
+  // actually get sent.
+  for (int i = 1; i <= 10; i++) {
+    caller()->data_channel()->Send(DataBuffer("Sent while blocked"));
+  }
+  // Nothing should be delivered during outage.
+  // We do a short wait to verify that delivery count is still 1.
+  WAIT(false, 10);
+  EXPECT_EQ(1u, callee()->data_observer()->received_message_count());
+  // Reverse the network outage.
+  virtual_socket_server()->set_drop_probability(0.0);
+  // Send a new packet, and wait for it to be delivered.
+  caller()->data_channel()->Send(DataBuffer("After block"));
+  EXPECT_EQ_WAIT("After block", callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  // Some messages should be lost, but first and last message should have
+  // been delivered.
+  // First, check that the protocol guarantee is preserved.
+  EXPECT_GT(11u, callee()->data_observer()->received_message_count());
+  EXPECT_LE(2u, callee()->data_observer()->received_message_count());
+  // Then, check that observed behavior (lose all messages) has not changed
+  EXPECT_EQ(2u, callee()->data_observer()->received_message_count());
+}
+
+TEST_P(DataChannelIntegrationTest,
+       QueuedPacketsGetDroppedInLifetimeLimitedMode) {
+  CreatePeerConnectionWrappers();
+  ConnectFakeSignaling();
+  DataChannelInit init;
+  init.maxRetransmitTime = 1;
+  init.ordered = false;
+  caller()->CreateDataChannel(&init);
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
+  caller()->data_channel()->Send(DataBuffer("hello first"));
+  ASSERT_EQ_WAIT(1u, callee()->data_observer()->received_message_count(),
+                 kDefaultTimeout);
+  // Cause a temporary network outage
+  virtual_socket_server()->set_drop_probability(1.0);
+  for (int i = 1; i <= 200; i++) {
+    caller()->data_channel()->Send(DataBuffer("Sent while blocked"));
+  }
+  // Nothing should be delivered during outage.
+  // We do a short wait to verify that delivery count is still 1,
+  // and to make sure max packet lifetime (which is in ms) is exceeded.
+  WAIT(false, 10);
+  EXPECT_EQ(1u, callee()->data_observer()->received_message_count());
+  // Reverse the network outage.
+  virtual_socket_server()->set_drop_probability(0.0);
+  // Send a new packet, and wait for it to be delivered.
+  caller()->data_channel()->Send(DataBuffer("After block"));
+  EXPECT_EQ_WAIT("After block", callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  // Some messages should be lost, but first and last message should have
+  // been delivered.
+  // First, check that the protocol guarantee is preserved.
+  EXPECT_GT(202u, callee()->data_observer()->received_message_count());
+  EXPECT_LE(2u, callee()->data_observer()->received_message_count());
+  // Then, check that observed behavior (lose some messages) has not changed
+  // DcSctp loses all messages. This is correct.
+  EXPECT_EQ(2u, callee()->data_observer()->received_message_count());
+}
+
+TEST_P(DataChannelIntegrationTest,
+       DISABLED_ON_ANDROID(SomeQueuedPacketsGetDroppedInMaxRetransmitsMode)) {
+  CreatePeerConnectionWrappers();
+  ConnectFakeSignaling();
+  DataChannelInit init;
+  init.maxRetransmits = 0;
+  init.ordered = false;
+  caller()->CreateDataChannel(&init);
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
+  caller()->data_channel()->Send(DataBuffer("hello first"));
+  ASSERT_EQ_WAIT(1u, callee()->data_observer()->received_message_count(),
+                 kDefaultTimeout);
+  // Cause a temporary network outage
+  virtual_socket_server()->set_drop_probability(1.0);
+  // Fill the buffer until queued data starts to build
+  size_t packet_counter = 0;
+  while (caller()->data_channel()->buffered_amount() < 1 &&
+         packet_counter < 10000) {
+    packet_counter++;
+    caller()->data_channel()->Send(DataBuffer("Sent while blocked"));
+  }
+  if (caller()->data_channel()->buffered_amount()) {
+    RTC_LOG(LS_INFO) << "Buffered data after " << packet_counter << " packets";
+  } else {
+    RTC_LOG(LS_INFO) << "No buffered data after " << packet_counter
+                     << " packets";
+  }
+  // Nothing should be delivered during outage.
+  // We do a short wait to verify that delivery count is still 1.
+  WAIT(false, 10);
+  EXPECT_EQ(1u, callee()->data_observer()->received_message_count());
+  // Reverse the network outage.
+  virtual_socket_server()->set_drop_probability(0.0);
+  // Send a new packet, and wait for it to be delivered.
+  caller()->data_channel()->Send(DataBuffer("After block"));
+  EXPECT_EQ_WAIT("After block", callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  // Some messages should be lost, but first and last message should have
+  // been delivered.
+  // Due to the fact that retransmissions are only counted when the packet
+  // goes on the wire, NOT when they are stalled in queue due to
+  // congestion, we expect some of the packets to be delivered, because
+  // congestion prevented them from being sent.
+  // Citation: https://tools.ietf.org/html/rfc7496#section-3.1
+
+  // First, check that the protocol guarantee is preserved.
+  EXPECT_GT(packet_counter,
+            callee()->data_observer()->received_message_count());
+  EXPECT_LE(2u, callee()->data_observer()->received_message_count());
+  // Then, check that observed behavior (lose between 100 and 200 messages)
+  // has not changed.
+  // Usrsctp behavior is different on Android (177) and other platforms (122).
+  // Dcsctp loses 432 packets.
+  EXPECT_GT(2 + packet_counter - 100,
+            callee()->data_observer()->received_message_count());
+  EXPECT_LT(2 + packet_counter - 500,
+            callee()->data_observer()->received_message_count());
+}
+
+INSTANTIATE_TEST_SUITE_P(DataChannelIntegrationTest,
+                         DataChannelIntegrationTest,
+                         Combine(Values(SdpSemantics::kPlanB_DEPRECATED,
+                                        SdpSemantics::kUnifiedPlan),
+                                 testing::Bool()));
+
+TEST_F(DataChannelIntegrationTestUnifiedPlan,
+       EndToEndCallWithBundledSctpDataChannel) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(caller()->pc()->GetSctpTransport(), kDefaultTimeout);
+  ASSERT_EQ_WAIT(SctpTransportState::kConnected,
+                 caller()->pc()->GetSctpTransport()->Information().state(),
+                 kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+}
+
+TEST_F(DataChannelIntegrationTestUnifiedPlan,
+       EndToEndCallWithDataChannelOnlyConnects) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+  ASSERT_TRUE(caller()->data_observer()->IsOpen());
+}
+
+TEST_F(DataChannelIntegrationTestUnifiedPlan, DataChannelClosesWhenClosed) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+  caller()->data_channel()->Close();
+  ASSERT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
+}
+
+TEST_F(DataChannelIntegrationTestUnifiedPlan,
+       DataChannelClosesWhenClosedReverse) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+  callee()->data_channel()->Close();
+  ASSERT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout);
+}
+
+TEST_F(DataChannelIntegrationTestUnifiedPlan,
+       DataChannelClosesWhenPeerConnectionClosed) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+  caller()->pc()->Close();
+  ASSERT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
+}
+
+#endif  // WEBRTC_HAVE_SCTP
+
+}  // namespace
+
+}  // namespace webrtc