| /* |
| * 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 "api/data_channel_interface.h" |
| #include "api/environment/environment.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/network/received_packet.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 DataMessageType::kControl: |
| return WebrtcPPID::kDCEP; |
| case DataMessageType::kText: |
| return size > 0 ? WebrtcPPID::kString : WebrtcPPID::kStringEmpty; |
| case 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 DataMessageType::kControl; |
| case WebrtcPPID::kString: |
| case WebrtcPPID::kStringPartial: |
| case WebrtcPPID::kStringEmpty: |
| return DataMessageType::kText; |
| case WebrtcPPID::kBinary: |
| case WebrtcPPID::kBinaryPartial: |
| case WebrtcPPID::kBinaryEmpty: |
| return 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(const Environment& env, |
| rtc::Thread* network_thread, |
| rtc::PacketTransportInternal* transport) |
| : DcSctpTransport(env, |
| network_thread, |
| transport, |
| std::make_unique<dcsctp::DcSctpSocketFactory>()) {} |
| DcSctpTransport::DcSctpTransport( |
| const Environment& env, |
| rtc::Thread* network_thread, |
| rtc::PacketTransportInternal* transport, |
| std::unique_ptr<dcsctp::DcSctpSocketFactory> socket_factory) |
| : network_thread_(network_thread), |
| transport_(transport), |
| env_(env), |
| random_(env_.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; |
| options.per_stream_send_queue_limit = |
| DataChannelInterface::MaxSendQueueSize(); |
| // This is just set to avoid denial-of-service. Practically unlimited. |
| options.max_send_buffer_size = std::numeric_limits<size_t>::max(); |
| |
| 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; |
| } |
| |
| RTCError DcSctpTransport::SendData(int sid, |
| const SendDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload) { |
| 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."; |
| return RTCError(RTCErrorType::INVALID_STATE); |
| } |
| |
| // 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; |
| return RTCError(RTCErrorType::INVALID_STATE); |
| } |
| |
| 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; |
| return RTCError(RTCErrorType::INVALID_STATE); |
| } |
| |
| 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; |
| return RTCError(RTCErrorType::INVALID_RANGE); |
| } |
| |
| 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; |
| } |
| |
| dcsctp::SendStatus error = socket_->Send(std::move(message), send_options); |
| switch (error) { |
| case dcsctp::SendStatus::kSuccess: |
| return RTCError::OK(); |
| case dcsctp::SendStatus::kErrorResourceExhaustion: |
| ready_to_send_data_ = false; |
| return RTCError(RTCErrorType::RESOURCE_EXHAUSTED); |
| default: |
| absl::string_view message = dcsctp::ToString(error); |
| RTC_LOG(LS_ERROR) << debug_name_ |
| << "->SendData(...): send() failed with error " |
| << message << "."; |
| return RTCError(RTCErrorType::NETWORK_ERROR, message); |
| } |
| } |
| |
| bool DcSctpTransport::ReadyToSendData() { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| 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; |
| } |
| |
| size_t DcSctpTransport::buffered_amount(int sid) const { |
| if (!socket_) |
| return 0; |
| return socket_->buffered_amount(dcsctp::StreamID(sid)); |
| } |
| |
| size_t DcSctpTransport::buffered_amount_low_threshold(int sid) const { |
| if (!socket_) |
| return 0; |
| return socket_->buffered_amount_low_threshold(dcsctp::StreamID(sid)); |
| } |
| |
| void DcSctpTransport::SetBufferedAmountLowThreshold(int sid, size_t bytes) { |
| if (!socket_) |
| return; |
| socket_->SetBufferedAmountLowThreshold(dcsctp::StreamID(sid), bytes); |
| } |
| |
| 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( |
| TaskQueueBase::DelayPrecision precision) { |
| return task_queue_timeout_factory_.CreateTimeout(precision); |
| } |
| |
| dcsctp::TimeMs DcSctpTransport::TimeMillis() { |
| return dcsctp::TimeMs(env_.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::OnBufferedAmountLow(dcsctp::StreamID stream_id) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (data_channel_sink_) { |
| data_channel_sink_->OnBufferedAmountLow(*stream_id); |
| } |
| } |
| |
| 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() << ")."; |
| 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."; |
| return; |
| } |
| receive_buffer_.Clear(); |
| if (!IsEmptyPPID(message.ppid())) |
| receive_buffer_.AppendData(message.payload().data(), |
| message.payload().size()); |
| |
| if (data_channel_sink_) { |
| data_channel_sink_->OnDataReceived(message.stream_id().value(), *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_DCHECK_RUN_ON(network_thread_); |
| 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_->RegisterReceivedPacketCallback( |
| this, [&](rtc::PacketTransportInternal* transport, |
| const rtc::ReceivedPacket& packet) { |
| OnTransportReadPacket(transport, packet); |
| }); |
| transport_->SetOnCloseCallback([this]() { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| RTC_DLOG(LS_VERBOSE) << debug_name_ << "->OnTransportClosed()."; |
| if (data_channel_sink_) { |
| data_channel_sink_->OnTransportClosed({}); |
| } |
| }); |
| } |
| |
| void DcSctpTransport::DisconnectTransportSignals() { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (!transport_) { |
| return; |
| } |
| transport_->SignalWritableState.disconnect(this); |
| transport_->DeregisterReceivedPacketCallback(this); |
| transport_->SetOnCloseCallback(nullptr); |
| } |
| |
| 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 rtc::ReceivedPacket& packet) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (packet.decryption_info() != rtc::ReceivedPacket::kDtlsDecrypted) { |
| // We are only interested in SCTP packets. |
| return; |
| } |
| |
| RTC_DLOG(LS_VERBOSE) << debug_name_ << "->OnTransportReadPacket(), length=" |
| << packet.payload().size(); |
| if (socket_) { |
| socket_->ReceivePacket(packet.payload()); |
| } |
| } |
| |
| void DcSctpTransport::MaybeConnectSocket() { |
| if (transport_ && transport_->writable() && socket_ && |
| socket_->state() == dcsctp::SocketState::kClosed) { |
| socket_->Connect(); |
| } |
| } |
| } // namespace webrtc |
