| /* |
| * 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 <algorithm> |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/data_channel_interface.h" |
| #include "api/dtmf_sender_interface.h" |
| #include "api/peer_connection_interface.h" |
| #include "api/scoped_refptr.h" |
| #include "api/units/time_delta.h" |
| #include "pc/test/integration_test_helpers.h" |
| #include "pc/test/mock_peer_connection_observers.h" |
| #include "rtc_base/fake_clock.h" |
| #include "rtc_base/gunit.h" |
| #include "rtc_base/ref_counted_object.h" |
| #include "rtc_base/virtual_socket_server.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| class DataChannelIntegrationTest |
| : public PeerConnectionIntegrationBaseTest, |
| public ::testing::WithParamInterface<SdpSemantics> { |
| protected: |
| DataChannelIntegrationTest() |
| : PeerConnectionIntegrationBaseTest(GetParam()) {} |
| }; |
| |
| GTEST_ALLOW_UNINSTANTIATED_PARAMETERIZED_TEST(DataChannelIntegrationTest); |
| |
| // 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; } |
| }; |
| |
| // Ensure FakeClockForTest is constructed first (see class for rationale). |
| class DataChannelIntegrationTestWithFakeClock |
| : public FakeClockForTest, |
| public DataChannelIntegrationTest {}; |
| |
| class DataChannelIntegrationTestPlanB |
| : public PeerConnectionIntegrationBaseTest { |
| protected: |
| DataChannelIntegrationTestPlanB() |
| : PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB) {} |
| }; |
| |
| GTEST_ALLOW_UNINSTANTIATED_PARAMETERIZED_TEST( |
| DataChannelIntegrationTestWithFakeClock); |
| |
| class DataChannelIntegrationTestUnifiedPlan |
| : public PeerConnectionIntegrationBaseTest { |
| protected: |
| DataChannelIntegrationTestUnifiedPlan() |
| : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {} |
| }; |
| |
| #ifdef WEBRTC_HAVE_SCTP |
| |
| // 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(); |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| // 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); |
| } |
| } |
| |
| 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(); |
| 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(); |
| EXPECT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout); |
| EXPECT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout); |
| } |
| |
| TEST_P(DataChannelIntegrationTest, SctpDataChannelConfigSentToOtherSide) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| webrtc::DataChannelInit init; |
| init.id = 53; |
| init.maxRetransmits = 52; |
| caller()->CreateDataChannel("data-channel", &init); |
| 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 usrsctp'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 = |
| caller()->data_observer()->messages(); |
| std::vector<std::string> callee_received_messages = |
| callee()->data_observer()->messages(); |
| 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); |
| } |
| |
| // 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) { |
| 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) { |
| 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); |
| } |
| |
| #endif // WEBRTC_HAVE_SCTP |
| |
| // Test that after closing PeerConnections, they stop sending any packets (ICE, |
| // DTLS, RTP...). |
| TEST_P(DataChannelIntegrationTest, ClosingConnectionStopsPacketFlow) { |
| // Set up audio/video/data, wait for some frames to be received. |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| #ifdef WEBRTC_HAVE_SCTP |
| caller()->CreateDataChannel(); |
| #endif |
| 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 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. |
| #ifdef WEBRTC_HAVE_SCTP |
| 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()); |
| } |
| |
| INSTANTIATE_TEST_SUITE_P(DataChannelIntegrationTest, |
| DataChannelIntegrationTest, |
| Values(SdpSemantics::kPlanB, |
| SdpSemantics::kUnifiedPlan)); |
| |
| INSTANTIATE_TEST_SUITE_P(DataChannelIntegrationTest, |
| DataChannelIntegrationTestWithFakeClock, |
| Values(SdpSemantics::kPlanB, |
| SdpSemantics::kUnifiedPlan)); |
| |
| 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 |