dtls_transport will when detecting a new fingerprint (e.g by usage of pranswer) signal DtlsTransportState::kNew. When this happen, the dtls crypto state is lost, and sctp should reconnect, srtp does this automatically in current code base. The existing behavior in dcsctp is that it will detect peer sending an init, and reconnect. But any messages sent between the dtls restart and the message arriving from the peer will be lost. This patch changes so that this case is gracefully handled by a) letting dcsctp_transport listen to dtls state this is big part of patch and involves changing the type of the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this into a separate patch... b) if a dtls restart happens, delete and restart socket. Testcase that fails before patch and works after is attached. Bonus: And include-what-you-use on patch Bug: b/375327137 Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43546}
228 lines
8.0 KiB
C++
228 lines
8.0 KiB
C++
/*
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* Copyright 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/sctp_transport.h"
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#include <cstddef>
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#include <functional>
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#include <memory>
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#include <optional>
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#include <utility>
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#include <vector>
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#include "absl/memory/memory.h"
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#include "api/dtls_transport_interface.h"
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#include "api/make_ref_counted.h"
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#include "api/priority.h"
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#include "api/rtc_error.h"
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#include "api/scoped_refptr.h"
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#include "api/sctp_transport_interface.h"
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#include "api/transport/data_channel_transport_interface.h"
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#include "media/sctp/sctp_transport_internal.h"
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#include "p2p/base/p2p_constants.h"
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#include "p2p/base/packet_transport_internal.h"
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#include "p2p/dtls/dtls_transport_internal.h"
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#include "p2p/dtls/fake_dtls_transport.h"
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#include "pc/dtls_transport.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/gunit.h"
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#include "rtc_base/thread.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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constexpr int kDefaultTimeout = 1000; // milliseconds
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constexpr int kTestMaxSctpStreams = 1234;
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using cricket::FakeDtlsTransport;
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using ::testing::ElementsAre;
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namespace webrtc {
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namespace {
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class FakeCricketSctpTransport : public cricket::SctpTransportInternal {
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public:
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void SetOnConnectedCallback(std::function<void()> callback) override {
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on_connected_callback_ = std::move(callback);
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}
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void SetDataChannelSink(DataChannelSink* sink) override {}
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void SetDtlsTransport(cricket::DtlsTransportInternal* transport) override {}
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bool Start(int local_port, int remote_port, int max_message_size) override {
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return true;
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}
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bool OpenStream(int sid, PriorityValue priority) override { return true; }
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bool ResetStream(int sid) override { return true; }
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RTCError SendData(int sid,
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const SendDataParams& params,
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const rtc::CopyOnWriteBuffer& payload) override {
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return RTCError::OK();
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}
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bool ReadyToSendData() override { return true; }
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void set_debug_name_for_testing(const char* debug_name) override {}
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int max_message_size() const override { return 0; }
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std::optional<int> max_outbound_streams() const override {
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return max_outbound_streams_;
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}
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std::optional<int> max_inbound_streams() const override {
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return max_inbound_streams_;
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}
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size_t buffered_amount(int sid) const override { return 0; }
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size_t buffered_amount_low_threshold(int sid) const override { return 0; }
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void SetBufferedAmountLowThreshold(int sid, size_t bytes) override {}
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void SendSignalAssociationChangeCommunicationUp() {
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ASSERT_TRUE(on_connected_callback_);
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on_connected_callback_();
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}
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void set_max_outbound_streams(int streams) {
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max_outbound_streams_ = streams;
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}
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void set_max_inbound_streams(int streams) { max_inbound_streams_ = streams; }
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private:
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std::optional<int> max_outbound_streams_;
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std::optional<int> max_inbound_streams_;
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std::function<void()> on_connected_callback_;
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};
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} // namespace
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class TestSctpTransportObserver : public SctpTransportObserverInterface {
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public:
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TestSctpTransportObserver() : info_(SctpTransportState::kNew) {}
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void OnStateChange(SctpTransportInformation info) override {
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info_ = info;
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states_.push_back(info.state());
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}
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SctpTransportState State() {
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if (states_.size() > 0) {
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return states_[states_.size() - 1];
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} else {
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return SctpTransportState::kNew;
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}
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}
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const std::vector<SctpTransportState>& States() { return states_; }
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SctpTransportInformation LastReceivedInformation() { return info_; }
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private:
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std::vector<SctpTransportState> states_;
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SctpTransportInformation info_;
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};
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class SctpTransportTest : public ::testing::Test {
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public:
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SctpTransport* transport() { return transport_.get(); }
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SctpTransportObserverInterface* observer() { return &observer_; }
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void CreateTransport() {
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std::unique_ptr<cricket::DtlsTransportInternal> cricket_transport =
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std::make_unique<FakeDtlsTransport>(
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"audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
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dtls_transport_ =
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rtc::make_ref_counted<DtlsTransport>(std::move(cricket_transport));
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auto cricket_sctp_transport =
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absl::WrapUnique(new FakeCricketSctpTransport());
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transport_ = rtc::make_ref_counted<SctpTransport>(
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std::move(cricket_sctp_transport), dtls_transport_);
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}
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void CompleteSctpHandshake() {
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// The computed MaxChannels shall be the minimum of the outgoing
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// and incoming # of streams.
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CricketSctpTransport()->set_max_outbound_streams(kTestMaxSctpStreams);
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CricketSctpTransport()->set_max_inbound_streams(kTestMaxSctpStreams + 1);
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CricketSctpTransport()->SendSignalAssociationChangeCommunicationUp();
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}
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FakeCricketSctpTransport* CricketSctpTransport() {
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return static_cast<FakeCricketSctpTransport*>(transport_->internal());
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}
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rtc::AutoThread main_thread_;
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rtc::scoped_refptr<SctpTransport> transport_;
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rtc::scoped_refptr<DtlsTransport> dtls_transport_;
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TestSctpTransportObserver observer_;
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};
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TEST(SctpTransportSimpleTest, CreateClearDelete) {
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rtc::AutoThread main_thread;
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std::unique_ptr<cricket::DtlsTransportInternal> cricket_transport =
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std::make_unique<FakeDtlsTransport>("audio",
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cricket::ICE_CANDIDATE_COMPONENT_RTP);
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rtc::scoped_refptr<DtlsTransport> dtls_transport =
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rtc::make_ref_counted<DtlsTransport>(std::move(cricket_transport));
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std::unique_ptr<cricket::SctpTransportInternal> fake_cricket_sctp_transport =
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absl::WrapUnique(new FakeCricketSctpTransport());
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rtc::scoped_refptr<SctpTransport> sctp_transport =
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rtc::make_ref_counted<SctpTransport>(
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std::move(fake_cricket_sctp_transport), dtls_transport);
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ASSERT_TRUE(sctp_transport->internal());
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ASSERT_EQ(SctpTransportState::kConnecting,
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sctp_transport->Information().state());
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sctp_transport->Clear();
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ASSERT_FALSE(sctp_transport->internal());
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ASSERT_EQ(SctpTransportState::kClosed, sctp_transport->Information().state());
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}
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TEST_F(SctpTransportTest, EventsObservedWhenConnecting) {
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CreateTransport();
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transport()->RegisterObserver(observer());
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CompleteSctpHandshake();
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ASSERT_EQ_WAIT(SctpTransportState::kConnected, observer_.State(),
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kDefaultTimeout);
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EXPECT_THAT(observer_.States(), ElementsAre(SctpTransportState::kConnected));
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}
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TEST_F(SctpTransportTest, CloseWhenClearing) {
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CreateTransport();
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transport()->RegisterObserver(observer());
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CompleteSctpHandshake();
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ASSERT_EQ_WAIT(SctpTransportState::kConnected, observer_.State(),
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kDefaultTimeout);
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transport()->Clear();
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ASSERT_EQ_WAIT(SctpTransportState::kClosed, observer_.State(),
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kDefaultTimeout);
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}
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TEST_F(SctpTransportTest, MaxChannelsSignalled) {
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CreateTransport();
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transport()->RegisterObserver(observer());
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EXPECT_FALSE(transport()->Information().MaxChannels());
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EXPECT_FALSE(observer_.LastReceivedInformation().MaxChannels());
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CompleteSctpHandshake();
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ASSERT_EQ_WAIT(SctpTransportState::kConnected, observer_.State(),
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kDefaultTimeout);
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EXPECT_TRUE(transport()->Information().MaxChannels());
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EXPECT_EQ(kTestMaxSctpStreams, *(transport()->Information().MaxChannels()));
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EXPECT_TRUE(observer_.LastReceivedInformation().MaxChannels());
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EXPECT_EQ(kTestMaxSctpStreams,
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*(observer_.LastReceivedInformation().MaxChannels()));
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}
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TEST_F(SctpTransportTest, CloseWhenTransportCloses) {
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CreateTransport();
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transport()->RegisterObserver(observer());
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CompleteSctpHandshake();
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ASSERT_EQ_WAIT(SctpTransportState::kConnected, observer_.State(),
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kDefaultTimeout);
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static_cast<cricket::FakeDtlsTransport*>(dtls_transport_->internal())
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->SetDtlsState(DtlsTransportState::kClosed);
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ASSERT_EQ_WAIT(SctpTransportState::kClosed, observer_.State(),
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kDefaultTimeout);
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}
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} // namespace webrtc
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