summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/media/sctp/dcsctp_transport.cc
blob: 062360d251b84de5d8ce46db8002a22d2cf72e66 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
/*
 *  Copyright 2021 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 "media/sctp/dcsctp_transport.h"

#include <atomic>
#include <cstdint>
#include <limits>
#include <utility>
#include <vector>

#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "media/base/media_channel.h"
#include "net/dcsctp/public/dcsctp_socket_factory.h"
#include "net/dcsctp/public/packet_observer.h"
#include "net/dcsctp/public/text_pcap_packet_observer.h"
#include "net/dcsctp/public/types.h"
#include "p2p/base/packet_transport_internal.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/socket.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/thread.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"

namespace webrtc {

namespace {
using ::dcsctp::SendPacketStatus;

// When there is packet loss for a long time, the SCTP retry timers will use
// exponential backoff, which can grow to very long durations and when the
// connection recovers, it may take a long time to reach the new backoff
// duration. By limiting it to a reasonable limit, the time to recover reduces.
constexpr dcsctp::DurationMs kMaxTimerBackoffDuration =
    dcsctp::DurationMs(3000);

enum class WebrtcPPID : dcsctp::PPID::UnderlyingType {
  // https://www.rfc-editor.org/rfc/rfc8832.html#section-8.1
  kDCEP = 50,
  // https://www.rfc-editor.org/rfc/rfc8831.html#section-8
  kString = 51,
  kBinaryPartial = 52,  // Deprecated
  kBinary = 53,
  kStringPartial = 54,  // Deprecated
  kStringEmpty = 56,
  kBinaryEmpty = 57,
};

WebrtcPPID ToPPID(DataMessageType message_type, size_t size) {
  switch (message_type) {
    case webrtc::DataMessageType::kControl:
      return WebrtcPPID::kDCEP;
    case webrtc::DataMessageType::kText:
      return size > 0 ? WebrtcPPID::kString : WebrtcPPID::kStringEmpty;
    case webrtc::DataMessageType::kBinary:
      return size > 0 ? WebrtcPPID::kBinary : WebrtcPPID::kBinaryEmpty;
  }
}

absl::optional<DataMessageType> ToDataMessageType(dcsctp::PPID ppid) {
  switch (static_cast<WebrtcPPID>(ppid.value())) {
    case WebrtcPPID::kDCEP:
      return webrtc::DataMessageType::kControl;
    case WebrtcPPID::kString:
    case WebrtcPPID::kStringPartial:
    case WebrtcPPID::kStringEmpty:
      return webrtc::DataMessageType::kText;
    case WebrtcPPID::kBinary:
    case WebrtcPPID::kBinaryPartial:
    case WebrtcPPID::kBinaryEmpty:
      return webrtc::DataMessageType::kBinary;
  }
  return absl::nullopt;
}

absl::optional<cricket::SctpErrorCauseCode> ToErrorCauseCode(
    dcsctp::ErrorKind error) {
  switch (error) {
    case dcsctp::ErrorKind::kParseFailed:
      return cricket::SctpErrorCauseCode::kUnrecognizedParameters;
    case dcsctp::ErrorKind::kPeerReported:
      return cricket::SctpErrorCauseCode::kUserInitiatedAbort;
    case dcsctp::ErrorKind::kWrongSequence:
    case dcsctp::ErrorKind::kProtocolViolation:
      return cricket::SctpErrorCauseCode::kProtocolViolation;
    case dcsctp::ErrorKind::kResourceExhaustion:
      return cricket::SctpErrorCauseCode::kOutOfResource;
    case dcsctp::ErrorKind::kTooManyRetries:
    case dcsctp::ErrorKind::kUnsupportedOperation:
    case dcsctp::ErrorKind::kNoError:
    case dcsctp::ErrorKind::kNotConnected:
      // No SCTP error cause code matches those
      break;
  }
  return absl::nullopt;
}

bool IsEmptyPPID(dcsctp::PPID ppid) {
  WebrtcPPID webrtc_ppid = static_cast<WebrtcPPID>(ppid.value());
  return webrtc_ppid == WebrtcPPID::kStringEmpty ||
         webrtc_ppid == WebrtcPPID::kBinaryEmpty;
}
}  // namespace

DcSctpTransport::DcSctpTransport(rtc::Thread* network_thread,
                                 rtc::PacketTransportInternal* transport,
                                 Clock* clock)
    : DcSctpTransport(network_thread,
                      transport,
                      clock,
                      std::make_unique<dcsctp::DcSctpSocketFactory>()) {}

DcSctpTransport::DcSctpTransport(
    rtc::Thread* network_thread,
    rtc::PacketTransportInternal* transport,
    Clock* clock,
    std::unique_ptr<dcsctp::DcSctpSocketFactory> socket_factory)
    : network_thread_(network_thread),
      transport_(transport),
      clock_(clock),
      random_(clock_->TimeInMicroseconds()),
      socket_factory_(std::move(socket_factory)),
      task_queue_timeout_factory_(
          *network_thread,
          [this]() { return TimeMillis(); },
          [this](dcsctp::TimeoutID timeout_id) {
            socket_->HandleTimeout(timeout_id);
          }) {
  RTC_DCHECK_RUN_ON(network_thread_);
  static std::atomic<int> instance_count = 0;
  rtc::StringBuilder sb;
  sb << debug_name_ << instance_count++;
  debug_name_ = sb.Release();
  ConnectTransportSignals();
}

DcSctpTransport::~DcSctpTransport() {
  if (socket_) {
    socket_->Close();
  }
}

void DcSctpTransport::SetOnConnectedCallback(std::function<void()> callback) {
  RTC_DCHECK_RUN_ON(network_thread_);
  on_connected_callback_ = std::move(callback);
}

void DcSctpTransport::SetDataChannelSink(DataChannelSink* sink) {
  RTC_DCHECK_RUN_ON(network_thread_);
  data_channel_sink_ = sink;
  if (data_channel_sink_ && ready_to_send_data_) {
    data_channel_sink_->OnReadyToSend();
  }
}

void DcSctpTransport::SetDtlsTransport(
    rtc::PacketTransportInternal* transport) {
  RTC_DCHECK_RUN_ON(network_thread_);
  DisconnectTransportSignals();
  transport_ = transport;
  ConnectTransportSignals();
  MaybeConnectSocket();
}

bool DcSctpTransport::Start(int local_sctp_port,
                            int remote_sctp_port,
                            int max_message_size) {
  RTC_DCHECK_RUN_ON(network_thread_);
  RTC_DCHECK(max_message_size > 0);
  RTC_DLOG(LS_INFO) << debug_name_ << "->Start(local=" << local_sctp_port
                    << ", remote=" << remote_sctp_port
                    << ", max_message_size=" << max_message_size << ")";

  if (!socket_) {
    dcsctp::DcSctpOptions options;
    options.local_port = local_sctp_port;
    options.remote_port = remote_sctp_port;
    options.max_message_size = max_message_size;
    options.max_timer_backoff_duration = kMaxTimerBackoffDuration;
    // Don't close the connection automatically on too many retransmissions.
    options.max_retransmissions = absl::nullopt;
    options.max_init_retransmits = absl::nullopt;

    std::unique_ptr<dcsctp::PacketObserver> packet_observer;
    if (RTC_LOG_CHECK_LEVEL(LS_VERBOSE)) {
      packet_observer =
          std::make_unique<dcsctp::TextPcapPacketObserver>(debug_name_);
    }

    socket_ = socket_factory_->Create(debug_name_, *this,
                                      std::move(packet_observer), options);
  } else {
    if (local_sctp_port != socket_->options().local_port ||
        remote_sctp_port != socket_->options().remote_port) {
      RTC_LOG(LS_ERROR)
          << debug_name_ << "->Start(local=" << local_sctp_port
          << ", remote=" << remote_sctp_port
          << "): Can't change ports on already started transport.";
      return false;
    }
    socket_->SetMaxMessageSize(max_message_size);
  }

  MaybeConnectSocket();

  return true;
}

bool DcSctpTransport::OpenStream(int sid) {
  RTC_DCHECK_RUN_ON(network_thread_);
  RTC_DLOG(LS_INFO) << debug_name_ << "->OpenStream(" << sid << ").";

  StreamState stream_state;
  stream_states_.insert_or_assign(dcsctp::StreamID(static_cast<uint16_t>(sid)),
                                  stream_state);
  return true;
}

bool DcSctpTransport::ResetStream(int sid) {
  RTC_DCHECK_RUN_ON(network_thread_);
  RTC_DLOG(LS_INFO) << debug_name_ << "->ResetStream(" << sid << ").";
  if (!socket_) {
    RTC_LOG(LS_ERROR) << debug_name_ << "->ResetStream(sid=" << sid
                      << "): Transport is not started.";
    return false;
  }

  dcsctp::StreamID streams[1] = {dcsctp::StreamID(static_cast<uint16_t>(sid))};

  auto it = stream_states_.find(streams[0]);
  if (it == stream_states_.end()) {
    RTC_LOG(LS_ERROR) << debug_name_ << "->ResetStream(sid=" << sid
                      << "): Stream is not open.";
    return false;
  }

  StreamState& stream_state = it->second;
  if (stream_state.closure_initiated || stream_state.incoming_reset_done ||
      stream_state.outgoing_reset_done) {
    // The closing procedure was already initiated by the remote, don't do
    // anything.
    return false;
  }
  stream_state.closure_initiated = true;
  socket_->ResetStreams(streams);
  return true;
}

bool DcSctpTransport::SendData(int sid,
                               const SendDataParams& params,
                               const rtc::CopyOnWriteBuffer& payload,
                               cricket::SendDataResult* result) {
  RTC_DCHECK_RUN_ON(network_thread_);
  RTC_DLOG(LS_VERBOSE) << debug_name_ << "->SendData(sid=" << sid
                       << ", type=" << static_cast<int>(params.type)
                       << ", length=" << payload.size() << ").";

  if (!socket_) {
    RTC_LOG(LS_ERROR) << debug_name_
                      << "->SendData(...): Transport is not started.";
    *result = cricket::SDR_ERROR;
    return false;
  }

  // It is possible for a message to be sent from the signaling thread at the
  // same time a data-channel is closing, but before the signaling thread is
  // aware of it. So we need to keep track of currently active data channels and
  // skip sending messages for the ones that are not open or closing.
  // The sending errors are not impacting the data channel API contract as
  // it is allowed to discard queued messages when the channel is closing.
  auto stream_state =
      stream_states_.find(dcsctp::StreamID(static_cast<uint16_t>(sid)));
  if (stream_state == stream_states_.end()) {
    RTC_LOG(LS_VERBOSE) << "Skipping message on non-open stream with sid: "
                        << sid;
    *result = cricket::SDR_ERROR;
    return false;
  }

  if (stream_state->second.closure_initiated ||
      stream_state->second.incoming_reset_done ||
      stream_state->second.outgoing_reset_done) {
    RTC_LOG(LS_VERBOSE) << "Skipping message on closing stream with sid: "
                        << sid;
    *result = cricket::SDR_ERROR;
    return false;
  }

  auto max_message_size = socket_->options().max_message_size;
  if (max_message_size > 0 && payload.size() > max_message_size) {
    RTC_LOG(LS_WARNING) << debug_name_
                        << "->SendData(...): "
                           "Trying to send packet bigger "
                           "than the max message size: "
                        << payload.size() << " vs max of " << max_message_size;
    *result = cricket::SDR_ERROR;
    return false;
  }

  std::vector<uint8_t> message_payload(payload.cdata(),
                                       payload.cdata() + payload.size());
  if (message_payload.empty()) {
    // https://www.rfc-editor.org/rfc/rfc8831.html#section-6.6
    // SCTP does not support the sending of empty user messages. Therefore, if
    // an empty message has to be sent, the appropriate PPID (WebRTC String
    // Empty or WebRTC Binary Empty) is used, and the SCTP user message of one
    // zero byte is sent.
    message_payload.push_back('\0');
  }

  dcsctp::DcSctpMessage message(
      dcsctp::StreamID(static_cast<uint16_t>(sid)),
      dcsctp::PPID(static_cast<uint16_t>(ToPPID(params.type, payload.size()))),
      std::move(message_payload));

  dcsctp::SendOptions send_options;
  send_options.unordered = dcsctp::IsUnordered(!params.ordered);
  if (params.max_rtx_ms.has_value()) {
    RTC_DCHECK(*params.max_rtx_ms >= 0 &&
               *params.max_rtx_ms <= std::numeric_limits<uint16_t>::max());
    send_options.lifetime = dcsctp::DurationMs(*params.max_rtx_ms);
  }
  if (params.max_rtx_count.has_value()) {
    RTC_DCHECK(*params.max_rtx_count >= 0 &&
               *params.max_rtx_count <= std::numeric_limits<uint16_t>::max());
    send_options.max_retransmissions = *params.max_rtx_count;
  }

  auto error = socket_->Send(std::move(message), send_options);
  switch (error) {
    case dcsctp::SendStatus::kSuccess:
      *result = cricket::SDR_SUCCESS;
      break;
    case dcsctp::SendStatus::kErrorResourceExhaustion:
      *result = cricket::SDR_BLOCK;
      ready_to_send_data_ = false;
      break;
    default:
      RTC_LOG(LS_ERROR) << debug_name_
                        << "->SendData(...): send() failed with error "
                        << dcsctp::ToString(error) << ".";
      *result = cricket::SDR_ERROR;
      break;
  }

  return *result == cricket::SDR_SUCCESS;
}

bool DcSctpTransport::ReadyToSendData() {
  return ready_to_send_data_;
}

int DcSctpTransport::max_message_size() const {
  if (!socket_) {
    RTC_LOG(LS_ERROR) << debug_name_
                      << "->max_message_size(...): Transport is not started.";
    return 0;
  }
  return socket_->options().max_message_size;
}

absl::optional<int> DcSctpTransport::max_outbound_streams() const {
  if (!socket_)
    return absl::nullopt;
  return socket_->options().announced_maximum_outgoing_streams;
}

absl::optional<int> DcSctpTransport::max_inbound_streams() const {
  if (!socket_)
    return absl::nullopt;
  return socket_->options().announced_maximum_incoming_streams;
}

void DcSctpTransport::set_debug_name_for_testing(const char* debug_name) {
  debug_name_ = debug_name;
}

SendPacketStatus DcSctpTransport::SendPacketWithStatus(
    rtc::ArrayView<const uint8_t> data) {
  RTC_DCHECK_RUN_ON(network_thread_);
  RTC_DCHECK(socket_);

  if (data.size() > (socket_->options().mtu)) {
    RTC_LOG(LS_ERROR) << debug_name_
                      << "->SendPacket(...): "
                         "SCTP seems to have made a packet that is bigger "
                         "than its official MTU: "
                      << data.size() << " vs max of " << socket_->options().mtu;
    return SendPacketStatus::kError;
  }
  TRACE_EVENT0("webrtc", "DcSctpTransport::SendPacket");

  if (!transport_ || !transport_->writable())
    return SendPacketStatus::kError;

  RTC_DLOG(LS_VERBOSE) << debug_name_ << "->SendPacket(length=" << data.size()
                       << ")";

  auto result =
      transport_->SendPacket(reinterpret_cast<const char*>(data.data()),
                             data.size(), rtc::PacketOptions(), 0);

  if (result < 0) {
    RTC_LOG(LS_WARNING) << debug_name_ << "->SendPacket(length=" << data.size()
                        << ") failed with error: " << transport_->GetError()
                        << ".";

    if (rtc::IsBlockingError(transport_->GetError())) {
      return SendPacketStatus::kTemporaryFailure;
    }
    return SendPacketStatus::kError;
  }
  return SendPacketStatus::kSuccess;
}

std::unique_ptr<dcsctp::Timeout> DcSctpTransport::CreateTimeout(
    webrtc::TaskQueueBase::DelayPrecision precision) {
  return task_queue_timeout_factory_.CreateTimeout(precision);
}

dcsctp::TimeMs DcSctpTransport::TimeMillis() {
  return dcsctp::TimeMs(clock_->TimeInMilliseconds());
}

uint32_t DcSctpTransport::GetRandomInt(uint32_t low, uint32_t high) {
  return random_.Rand(low, high);
}

void DcSctpTransport::OnTotalBufferedAmountLow() {
  RTC_DCHECK_RUN_ON(network_thread_);
  if (!ready_to_send_data_) {
    ready_to_send_data_ = true;
    if (data_channel_sink_) {
      data_channel_sink_->OnReadyToSend();
    }
  }
}

void DcSctpTransport::OnMessageReceived(dcsctp::DcSctpMessage message) {
  RTC_DCHECK_RUN_ON(network_thread_);
  RTC_DLOG(LS_VERBOSE) << debug_name_ << "->OnMessageReceived(sid="
                       << message.stream_id().value()
                       << ", ppid=" << message.ppid().value()
                       << ", length=" << message.payload().size() << ").";
  cricket::ReceiveDataParams receive_data_params;
  receive_data_params.sid = message.stream_id().value();
  auto type = ToDataMessageType(message.ppid());
  if (!type.has_value()) {
    RTC_LOG(LS_VERBOSE) << debug_name_
                        << "->OnMessageReceived(): Received an unknown PPID "
                        << message.ppid().value()
                        << " on an SCTP packet. Dropping.";
  }
  receive_data_params.type = *type;
  // No seq_num available from dcSCTP
  receive_data_params.seq_num = 0;
  receive_buffer_.Clear();
  if (!IsEmptyPPID(message.ppid()))
    receive_buffer_.AppendData(message.payload().data(),
                               message.payload().size());

  if (data_channel_sink_) {
    data_channel_sink_->OnDataReceived(
        receive_data_params.sid, receive_data_params.type, receive_buffer_);
  }
}

void DcSctpTransport::OnError(dcsctp::ErrorKind error,
                              absl::string_view message) {
  if (error == dcsctp::ErrorKind::kResourceExhaustion) {
    // Indicates that a message failed to be enqueued, because the send buffer
    // is full, which is a very common (and wanted) state for high throughput
    // sending/benchmarks.
    RTC_LOG(LS_VERBOSE) << debug_name_
                        << "->OnError(error=" << dcsctp::ToString(error)
                        << ", message=" << message << ").";
  } else {
    RTC_LOG(LS_ERROR) << debug_name_
                      << "->OnError(error=" << dcsctp::ToString(error)
                      << ", message=" << message << ").";
  }
}

void DcSctpTransport::OnAborted(dcsctp::ErrorKind error,
                                absl::string_view message) {
  RTC_DCHECK_RUN_ON(network_thread_);
  RTC_LOG(LS_ERROR) << debug_name_
                    << "->OnAborted(error=" << dcsctp::ToString(error)
                    << ", message=" << message << ").";
  ready_to_send_data_ = false;
  RTCError rtc_error(RTCErrorType::OPERATION_ERROR_WITH_DATA,
                     std::string(message));
  rtc_error.set_error_detail(RTCErrorDetailType::SCTP_FAILURE);
  auto code = ToErrorCauseCode(error);
  if (code.has_value()) {
    rtc_error.set_sctp_cause_code(static_cast<uint16_t>(*code));
  }
  if (data_channel_sink_) {
    data_channel_sink_->OnTransportClosed(rtc_error);
  }
}

void DcSctpTransport::OnConnected() {
  RTC_DCHECK_RUN_ON(network_thread_);
  RTC_DLOG(LS_INFO) << debug_name_ << "->OnConnected().";
  ready_to_send_data_ = true;
  if (data_channel_sink_) {
    data_channel_sink_->OnReadyToSend();
  }
  if (on_connected_callback_) {
    on_connected_callback_();
  }
}

void DcSctpTransport::OnClosed() {
  RTC_DLOG(LS_INFO) << debug_name_ << "->OnClosed().";
  ready_to_send_data_ = false;
}

void DcSctpTransport::OnConnectionRestarted() {
  RTC_DLOG(LS_INFO) << debug_name_ << "->OnConnectionRestarted().";
}

void DcSctpTransport::OnStreamsResetFailed(
    rtc::ArrayView<const dcsctp::StreamID> outgoing_streams,
    absl::string_view reason) {
  // TODO(orphis): Need a test to check for correct behavior
  for (auto& stream_id : outgoing_streams) {
    RTC_LOG(LS_WARNING)
        << debug_name_
        << "->OnStreamsResetFailed(...): Outgoing stream reset failed"
        << ", sid=" << stream_id.value() << ", reason: " << reason << ".";
  }
}

void DcSctpTransport::OnStreamsResetPerformed(
    rtc::ArrayView<const dcsctp::StreamID> outgoing_streams) {
  RTC_DCHECK_RUN_ON(network_thread_);
  for (auto& stream_id : outgoing_streams) {
    RTC_LOG(LS_INFO) << debug_name_
                     << "->OnStreamsResetPerformed(...): Outgoing stream reset"
                     << ", sid=" << stream_id.value();

    auto it = stream_states_.find(stream_id);
    if (it == stream_states_.end()) {
      // Ignoring an outgoing stream reset for a closed stream
      return;
    }

    StreamState& stream_state = it->second;
    stream_state.outgoing_reset_done = true;

    if (stream_state.incoming_reset_done) {
      //  When the close was not initiated locally, we can signal the end of the
      //  data channel close procedure when the remote ACKs the reset.
      if (data_channel_sink_) {
        data_channel_sink_->OnChannelClosed(stream_id.value());
      }
      stream_states_.erase(stream_id);
    }
  }
}

void DcSctpTransport::OnIncomingStreamsReset(
    rtc::ArrayView<const dcsctp::StreamID> incoming_streams) {
  RTC_DCHECK_RUN_ON(network_thread_);
  for (auto& stream_id : incoming_streams) {
    RTC_LOG(LS_INFO) << debug_name_
                     << "->OnIncomingStreamsReset(...): Incoming stream reset"
                     << ", sid=" << stream_id.value();

    auto it = stream_states_.find(stream_id);
    if (it == stream_states_.end())
      return;

    StreamState& stream_state = it->second;
    stream_state.incoming_reset_done = true;

    if (!stream_state.closure_initiated) {
      // When receiving an incoming stream reset event for a non local close
      // procedure, the transport needs to reset the stream in the other
      // direction too.
      dcsctp::StreamID streams[1] = {stream_id};
      socket_->ResetStreams(streams);
      if (data_channel_sink_) {
        data_channel_sink_->OnChannelClosing(stream_id.value());
      }
    }

    if (stream_state.outgoing_reset_done) {
      // The close procedure that was initiated locally is complete when we
      // receive and incoming reset event.
      if (data_channel_sink_) {
        data_channel_sink_->OnChannelClosed(stream_id.value());
      }
      stream_states_.erase(stream_id);
    }
  }
}

void DcSctpTransport::ConnectTransportSignals() {
  RTC_DCHECK_RUN_ON(network_thread_);
  if (!transport_) {
    return;
  }
  transport_->SignalWritableState.connect(
      this, &DcSctpTransport::OnTransportWritableState);
  transport_->SignalReadPacket.connect(this,
                                       &DcSctpTransport::OnTransportReadPacket);
  transport_->SignalClosed.connect(this, &DcSctpTransport::OnTransportClosed);
}

void DcSctpTransport::DisconnectTransportSignals() {
  RTC_DCHECK_RUN_ON(network_thread_);
  if (!transport_) {
    return;
  }
  transport_->SignalWritableState.disconnect(this);
  transport_->SignalReadPacket.disconnect(this);
  transport_->SignalClosed.disconnect(this);
}

void DcSctpTransport::OnTransportWritableState(
    rtc::PacketTransportInternal* transport) {
  RTC_DCHECK_RUN_ON(network_thread_);
  RTC_DCHECK_EQ(transport_, transport);
  RTC_DLOG(LS_VERBOSE) << debug_name_
                       << "->OnTransportWritableState(), writable="
                       << transport->writable();
  MaybeConnectSocket();
}

void DcSctpTransport::OnTransportReadPacket(
    rtc::PacketTransportInternal* transport,
    const char* data,
    size_t length,
    const int64_t& /* packet_time_us */,
    int flags) {
  RTC_DCHECK_RUN_ON(network_thread_);
  if (flags) {
    // We are only interested in SCTP packets.
    return;
  }

  RTC_DLOG(LS_VERBOSE) << debug_name_
                       << "->OnTransportReadPacket(), length=" << length;
  if (socket_) {
    socket_->ReceivePacket(rtc::ArrayView<const uint8_t>(
        reinterpret_cast<const uint8_t*>(data), length));
  }
}

void DcSctpTransport::OnTransportClosed(
    rtc::PacketTransportInternal* transport) {
  RTC_DCHECK_RUN_ON(network_thread_);
  RTC_DLOG(LS_VERBOSE) << debug_name_ << "->OnTransportClosed().";
  if (data_channel_sink_) {
    data_channel_sink_->OnTransportClosed({});
  }
}

void DcSctpTransport::MaybeConnectSocket() {
  if (transport_ && transport_->writable() && socket_ &&
      socket_->state() == dcsctp::SocketState::kClosed) {
    socket_->Connect();
  }
}
}  // namespace webrtc