This code looked a bit weird before this CL - probably because of old refactorings. In JsepTransport constructor, there is a DCHECK assuring that the RTP DTLS transport is always present, so it can be passed directly to the SctpTransport constructor, which avoids having the SetDtlsTransport method in it. Also, in the SctpTransport constructor, there was code that would set the SCTP transport state to `kConnecting` if the DTLS transport was present, but that was dead code, as it was always `nullptr` inside the constructor before this CL. With this CL, it's always present, and the SCTP Transport's state will initially always be `kConnecting` now. Which is a step to deprecating the `kNew` state that doesn't exist in https://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate. One test case was modified, as it didn't test the reality. The test created a SctpTransport, registered an observer, and added the DTLS transport, and expected to receive a "statechange" from `kNew` (which is not a state that exists in the spec) to `kConnecting`. If the test had tested the opposite ordering - adding the DTLS transport first, and then adding an observer, it wouldn't have experienced this. And since in reality (with the implementation of JsepTransport before and after this CL), it always adds the DTLS transport before any observer is registered. So it wouldn't ever be fired, outside of tests. Bug: webrtc:15897 Change-Id: I6ac24e0a331b686eb400fcf388ece50f2ad46a32 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345420 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41987}
204 lines
7.0 KiB
C++
204 lines
7.0 KiB
C++
/*
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* Copyright 2018 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 <algorithm>
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#include <utility>
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#include "absl/types/optional.h"
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#include "api/dtls_transport_interface.h"
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#include "api/sequence_checker.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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SctpTransport::SctpTransport(
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std::unique_ptr<cricket::SctpTransportInternal> internal,
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rtc::scoped_refptr<DtlsTransport> dtls_transport)
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: owner_thread_(rtc::Thread::Current()),
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info_(SctpTransportState::kConnecting,
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dtls_transport,
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/*max_message_size=*/absl::nullopt,
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/*max_channels=*/absl::nullopt),
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internal_sctp_transport_(std::move(internal)),
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dtls_transport_(dtls_transport) {
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RTC_DCHECK(internal_sctp_transport_.get());
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RTC_DCHECK(dtls_transport_.get());
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dtls_transport_->internal()->SubscribeDtlsTransportState(
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[this](cricket::DtlsTransportInternal* transport,
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DtlsTransportState state) {
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OnDtlsStateChange(transport, state);
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});
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internal_sctp_transport_->SetDtlsTransport(dtls_transport->internal());
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internal_sctp_transport_->SetOnConnectedCallback(
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[this]() { OnAssociationChangeCommunicationUp(); });
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}
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SctpTransport::~SctpTransport() {
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// We depend on the network thread to call Clear() before dropping
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// its last reference to this object.
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RTC_DCHECK(owner_thread_->IsCurrent() || !internal_sctp_transport_);
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}
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SctpTransportInformation SctpTransport::Information() const {
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// TODO(tommi): Update PeerConnection::GetSctpTransport to hand out a proxy
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// to the transport so that we can be sure that methods get called on the
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// expected thread. Chromium currently calls this method from
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// TransceiverStateSurfacer.
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if (!owner_thread_->IsCurrent()) {
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return owner_thread_->BlockingCall([this] { return Information(); });
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}
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RTC_DCHECK_RUN_ON(owner_thread_);
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return info_;
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}
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void SctpTransport::RegisterObserver(SctpTransportObserverInterface* observer) {
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RTC_DCHECK_RUN_ON(owner_thread_);
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RTC_DCHECK(observer);
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RTC_DCHECK(!observer_);
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observer_ = observer;
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}
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void SctpTransport::UnregisterObserver() {
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RTC_DCHECK_RUN_ON(owner_thread_);
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observer_ = nullptr;
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}
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RTCError SctpTransport::OpenChannel(int channel_id) {
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RTC_DCHECK_RUN_ON(owner_thread_);
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RTC_DCHECK(internal_sctp_transport_);
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internal_sctp_transport_->OpenStream(channel_id);
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return RTCError::OK();
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}
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RTCError SctpTransport::SendData(int channel_id,
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const SendDataParams& params,
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const rtc::CopyOnWriteBuffer& buffer) {
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RTC_DCHECK_RUN_ON(owner_thread_);
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return internal_sctp_transport_->SendData(channel_id, params, buffer);
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}
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RTCError SctpTransport::CloseChannel(int channel_id) {
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RTC_DCHECK_RUN_ON(owner_thread_);
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RTC_DCHECK(internal_sctp_transport_);
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internal_sctp_transport_->ResetStream(channel_id);
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return RTCError::OK();
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}
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void SctpTransport::SetDataSink(DataChannelSink* sink) {
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RTC_DCHECK_RUN_ON(owner_thread_);
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RTC_DCHECK(internal_sctp_transport_);
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internal_sctp_transport_->SetDataChannelSink(sink);
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}
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bool SctpTransport::IsReadyToSend() const {
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RTC_DCHECK_RUN_ON(owner_thread_);
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RTC_DCHECK(internal_sctp_transport_);
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return internal_sctp_transport_->ReadyToSendData();
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}
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size_t SctpTransport::buffered_amount(int channel_id) const {
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RTC_DCHECK_RUN_ON(owner_thread_);
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RTC_DCHECK(internal_sctp_transport_);
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return internal_sctp_transport_->buffered_amount(channel_id);
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}
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size_t SctpTransport::buffered_amount_low_threshold(int channel_id) const {
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RTC_DCHECK_RUN_ON(owner_thread_);
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return internal_sctp_transport_->buffered_amount_low_threshold(channel_id);
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}
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void SctpTransport::SetBufferedAmountLowThreshold(int channel_id,
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size_t bytes) {
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RTC_DCHECK_RUN_ON(owner_thread_);
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internal_sctp_transport_->SetBufferedAmountLowThreshold(channel_id, bytes);
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}
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rtc::scoped_refptr<DtlsTransportInterface> SctpTransport::dtls_transport()
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const {
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RTC_DCHECK_RUN_ON(owner_thread_);
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return dtls_transport_;
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}
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// Internal functions
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void SctpTransport::Clear() {
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RTC_DCHECK_RUN_ON(owner_thread_);
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RTC_DCHECK(internal());
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// Note that we delete internal_sctp_transport_, but
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// only drop the reference to dtls_transport_.
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dtls_transport_ = nullptr;
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internal_sctp_transport_ = nullptr;
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UpdateInformation(SctpTransportState::kClosed);
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}
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void SctpTransport::Start(int local_port,
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int remote_port,
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int max_message_size) {
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RTC_DCHECK_RUN_ON(owner_thread_);
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info_ = SctpTransportInformation(info_.state(), info_.dtls_transport(),
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max_message_size, info_.MaxChannels());
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if (!internal()->Start(local_port, remote_port, max_message_size)) {
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RTC_LOG(LS_ERROR) << "Failed to push down SCTP parameters, closing.";
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UpdateInformation(SctpTransportState::kClosed);
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}
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}
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void SctpTransport::UpdateInformation(SctpTransportState state) {
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RTC_DCHECK_RUN_ON(owner_thread_);
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bool must_send_update = (state != info_.state());
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// TODO(https://bugs.webrtc.org/10358): Update max channels from internal
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// SCTP transport when available.
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if (internal_sctp_transport_) {
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info_ = SctpTransportInformation(
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state, dtls_transport_, info_.MaxMessageSize(), info_.MaxChannels());
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} else {
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info_ = SctpTransportInformation(
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state, dtls_transport_, info_.MaxMessageSize(), info_.MaxChannels());
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}
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if (observer_ && must_send_update) {
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observer_->OnStateChange(info_);
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}
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}
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void SctpTransport::OnAssociationChangeCommunicationUp() {
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RTC_DCHECK_RUN_ON(owner_thread_);
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RTC_DCHECK(internal_sctp_transport_);
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if (internal_sctp_transport_->max_outbound_streams() &&
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internal_sctp_transport_->max_inbound_streams()) {
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int max_channels =
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std::min(*(internal_sctp_transport_->max_outbound_streams()),
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*(internal_sctp_transport_->max_inbound_streams()));
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// Record max channels.
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info_ = SctpTransportInformation(info_.state(), info_.dtls_transport(),
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info_.MaxMessageSize(), max_channels);
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}
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UpdateInformation(SctpTransportState::kConnected);
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}
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void SctpTransport::OnDtlsStateChange(cricket::DtlsTransportInternal* transport,
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DtlsTransportState state) {
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RTC_DCHECK_RUN_ON(owner_thread_);
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RTC_CHECK(transport == dtls_transport_->internal());
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if (state == DtlsTransportState::kClosed ||
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state == DtlsTransportState::kFailed) {
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UpdateInformation(SctpTransportState::kClosed);
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// TODO(http://bugs.webrtc.org/11090): Close all the data channels
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}
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}
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} // namespace webrtc
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