| /* |
| * Copyright 2020 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 "pc/sctp_data_channel.h" |
| |
| #include <limits> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| |
| #include "media/sctp/sctp_transport_internal.h" |
| #include "pc/proxy.h" |
| #include "pc/sctp_utils.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/ref_counted_object.h" |
| #include "rtc_base/task_utils/to_queued_task.h" |
| #include "rtc_base/thread.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| static size_t kMaxQueuedReceivedDataBytes = 16 * 1024 * 1024; |
| static size_t kMaxQueuedSendDataBytes = 16 * 1024 * 1024; |
| |
| static std::atomic<int> g_unique_id{0}; |
| |
| int GenerateUniqueId() { |
| return ++g_unique_id; |
| } |
| |
| // Define proxy for DataChannelInterface. |
| BEGIN_PRIMARY_PROXY_MAP(DataChannel) |
| PROXY_PRIMARY_THREAD_DESTRUCTOR() |
| PROXY_METHOD1(void, RegisterObserver, DataChannelObserver*) |
| PROXY_METHOD0(void, UnregisterObserver) |
| BYPASS_PROXY_CONSTMETHOD0(std::string, label) |
| BYPASS_PROXY_CONSTMETHOD0(bool, reliable) |
| BYPASS_PROXY_CONSTMETHOD0(bool, ordered) |
| BYPASS_PROXY_CONSTMETHOD0(uint16_t, maxRetransmitTime) |
| BYPASS_PROXY_CONSTMETHOD0(uint16_t, maxRetransmits) |
| BYPASS_PROXY_CONSTMETHOD0(absl::optional<int>, maxRetransmitsOpt) |
| BYPASS_PROXY_CONSTMETHOD0(absl::optional<int>, maxPacketLifeTime) |
| BYPASS_PROXY_CONSTMETHOD0(std::string, protocol) |
| BYPASS_PROXY_CONSTMETHOD0(bool, negotiated) |
| // Can't bypass the proxy since the id may change. |
| PROXY_CONSTMETHOD0(int, id) |
| BYPASS_PROXY_CONSTMETHOD0(Priority, priority) |
| PROXY_CONSTMETHOD0(DataState, state) |
| PROXY_CONSTMETHOD0(RTCError, error) |
| PROXY_CONSTMETHOD0(uint32_t, messages_sent) |
| PROXY_CONSTMETHOD0(uint64_t, bytes_sent) |
| PROXY_CONSTMETHOD0(uint32_t, messages_received) |
| PROXY_CONSTMETHOD0(uint64_t, bytes_received) |
| PROXY_CONSTMETHOD0(uint64_t, buffered_amount) |
| PROXY_METHOD0(void, Close) |
| // TODO(bugs.webrtc.org/11547): Change to run on the network thread. |
| PROXY_METHOD1(bool, Send, const DataBuffer&) |
| END_PROXY_MAP(DataChannel) |
| |
| } // namespace |
| |
| InternalDataChannelInit::InternalDataChannelInit(const DataChannelInit& base) |
| : DataChannelInit(base), open_handshake_role(kOpener) { |
| // If the channel is externally negotiated, do not send the OPEN message. |
| if (base.negotiated) { |
| open_handshake_role = kNone; |
| } else { |
| // Datachannel is externally negotiated. Ignore the id value. |
| // Specified in createDataChannel, WebRTC spec section 6.1 bullet 13. |
| id = -1; |
| } |
| // Backwards compatibility: If maxRetransmits or maxRetransmitTime |
| // are negative, the feature is not enabled. |
| // Values are clamped to a 16bit range. |
| if (maxRetransmits) { |
| if (*maxRetransmits < 0) { |
| RTC_LOG(LS_ERROR) |
| << "Accepting maxRetransmits < 0 for backwards compatibility"; |
| maxRetransmits = absl::nullopt; |
| } else if (*maxRetransmits > std::numeric_limits<uint16_t>::max()) { |
| maxRetransmits = std::numeric_limits<uint16_t>::max(); |
| } |
| } |
| |
| if (maxRetransmitTime) { |
| if (*maxRetransmitTime < 0) { |
| RTC_LOG(LS_ERROR) |
| << "Accepting maxRetransmitTime < 0 for backwards compatibility"; |
| maxRetransmitTime = absl::nullopt; |
| } else if (*maxRetransmitTime > std::numeric_limits<uint16_t>::max()) { |
| maxRetransmitTime = std::numeric_limits<uint16_t>::max(); |
| } |
| } |
| } |
| |
| bool SctpSidAllocator::AllocateSid(rtc::SSLRole role, int* sid) { |
| int potential_sid = (role == rtc::SSL_CLIENT) ? 0 : 1; |
| while (!IsSidAvailable(potential_sid)) { |
| potential_sid += 2; |
| if (potential_sid > static_cast<int>(cricket::kMaxSctpSid)) { |
| return false; |
| } |
| } |
| |
| *sid = potential_sid; |
| used_sids_.insert(potential_sid); |
| return true; |
| } |
| |
| bool SctpSidAllocator::ReserveSid(int sid) { |
| if (!IsSidAvailable(sid)) { |
| return false; |
| } |
| used_sids_.insert(sid); |
| return true; |
| } |
| |
| void SctpSidAllocator::ReleaseSid(int sid) { |
| auto it = used_sids_.find(sid); |
| if (it != used_sids_.end()) { |
| used_sids_.erase(it); |
| } |
| } |
| |
| bool SctpSidAllocator::IsSidAvailable(int sid) const { |
| if (sid < static_cast<int>(cricket::kMinSctpSid) || |
| sid > static_cast<int>(cricket::kMaxSctpSid)) { |
| return false; |
| } |
| return used_sids_.find(sid) == used_sids_.end(); |
| } |
| |
| rtc::scoped_refptr<SctpDataChannel> SctpDataChannel::Create( |
| SctpDataChannelProviderInterface* provider, |
| const std::string& label, |
| const InternalDataChannelInit& config, |
| rtc::Thread* signaling_thread, |
| rtc::Thread* network_thread) { |
| auto channel = rtc::make_ref_counted<SctpDataChannel>( |
| config, provider, label, signaling_thread, network_thread); |
| if (!channel->Init()) { |
| return nullptr; |
| } |
| return channel; |
| } |
| |
| // static |
| rtc::scoped_refptr<DataChannelInterface> SctpDataChannel::CreateProxy( |
| rtc::scoped_refptr<SctpDataChannel> channel) { |
| // TODO(bugs.webrtc.org/11547): incorporate the network thread in the proxy. |
| // Also, consider allowing the proxy object to own the reference (std::move). |
| // As is, the proxy has a raw pointer and no reference to the channel object |
| // and trusting that the lifetime management aligns with the |
| // sctp_data_channels_ array in SctpDataChannelController. |
| return DataChannelProxy::Create(channel->signaling_thread_, channel.get()); |
| } |
| |
| SctpDataChannel::SctpDataChannel(const InternalDataChannelInit& config, |
| SctpDataChannelProviderInterface* provider, |
| const std::string& label, |
| rtc::Thread* signaling_thread, |
| rtc::Thread* network_thread) |
| : signaling_thread_(signaling_thread), |
| network_thread_(network_thread), |
| internal_id_(GenerateUniqueId()), |
| label_(label), |
| config_(config), |
| observer_(nullptr), |
| provider_(provider) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| } |
| |
| bool SctpDataChannel::Init() { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| if (config_.id < -1 || |
| (config_.maxRetransmits && *config_.maxRetransmits < 0) || |
| (config_.maxRetransmitTime && *config_.maxRetransmitTime < 0)) { |
| RTC_LOG(LS_ERROR) << "Failed to initialize the SCTP data channel due to " |
| "invalid DataChannelInit."; |
| return false; |
| } |
| if (config_.maxRetransmits && config_.maxRetransmitTime) { |
| RTC_LOG(LS_ERROR) |
| << "maxRetransmits and maxRetransmitTime should not be both set."; |
| return false; |
| } |
| |
| switch (config_.open_handshake_role) { |
| case webrtc::InternalDataChannelInit::kNone: // pre-negotiated |
| handshake_state_ = kHandshakeReady; |
| break; |
| case webrtc::InternalDataChannelInit::kOpener: |
| handshake_state_ = kHandshakeShouldSendOpen; |
| break; |
| case webrtc::InternalDataChannelInit::kAcker: |
| handshake_state_ = kHandshakeShouldSendAck; |
| break; |
| } |
| |
| // Try to connect to the transport in case the transport channel already |
| // exists. |
| OnTransportChannelCreated(); |
| |
| // Checks if the transport is ready to send because the initial channel |
| // ready signal may have been sent before the DataChannel creation. |
| // This has to be done async because the upper layer objects (e.g. |
| // Chrome glue and WebKit) are not wired up properly until after this |
| // function returns. |
| if (provider_->ReadyToSendData()) { |
| AddRef(); |
| rtc::Thread::Current()->PostTask(ToQueuedTask( |
| [this] { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| if (state_ != kClosed) |
| OnTransportReady(true); |
| }, |
| [this] { Release(); })); |
| } |
| |
| return true; |
| } |
| |
| SctpDataChannel::~SctpDataChannel() { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| } |
| |
| void SctpDataChannel::RegisterObserver(DataChannelObserver* observer) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| observer_ = observer; |
| DeliverQueuedReceivedData(); |
| } |
| |
| void SctpDataChannel::UnregisterObserver() { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| observer_ = nullptr; |
| } |
| |
| bool SctpDataChannel::reliable() const { |
| // May be called on any thread. |
| return !config_.maxRetransmits && !config_.maxRetransmitTime; |
| } |
| |
| uint64_t SctpDataChannel::buffered_amount() const { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| return buffered_amount_; |
| } |
| |
| void SctpDataChannel::Close() { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| if (state_ == kClosed) |
| return; |
| SetState(kClosing); |
| // Will send queued data before beginning the underlying closing procedure. |
| UpdateState(); |
| } |
| |
| SctpDataChannel::DataState SctpDataChannel::state() const { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| return state_; |
| } |
| |
| RTCError SctpDataChannel::error() const { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| return error_; |
| } |
| |
| uint32_t SctpDataChannel::messages_sent() const { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| return messages_sent_; |
| } |
| |
| uint64_t SctpDataChannel::bytes_sent() const { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| return bytes_sent_; |
| } |
| |
| uint32_t SctpDataChannel::messages_received() const { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| return messages_received_; |
| } |
| |
| uint64_t SctpDataChannel::bytes_received() const { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| return bytes_received_; |
| } |
| |
| bool SctpDataChannel::Send(const DataBuffer& buffer) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| // TODO(bugs.webrtc.org/11547): Expect this method to be called on the network |
| // thread. Bring buffer management etc to the network thread and keep the |
| // operational state management on the signaling thread. |
| |
| if (state_ != kOpen) { |
| return false; |
| } |
| |
| buffered_amount_ += buffer.size(); |
| |
| // If the queue is non-empty, we're waiting for SignalReadyToSend, |
| // so just add to the end of the queue and keep waiting. |
| if (!queued_send_data_.Empty()) { |
| if (!QueueSendDataMessage(buffer)) { |
| RTC_LOG(LS_ERROR) << "Closing the DataChannel due to a failure to queue " |
| "additional data."; |
| // https://w3c.github.io/webrtc-pc/#dom-rtcdatachannel-send step 5 |
| // Note that the spec doesn't explicitly say to close in this situation. |
| CloseAbruptlyWithError(RTCError(RTCErrorType::RESOURCE_EXHAUSTED, |
| "Unable to queue data for sending")); |
| } |
| return true; |
| } |
| |
| SendDataMessage(buffer, true); |
| |
| // Always return true for SCTP DataChannel per the spec. |
| return true; |
| } |
| |
| void SctpDataChannel::SetSctpSid(int sid) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| RTC_DCHECK_LT(config_.id, 0); |
| RTC_DCHECK_GE(sid, 0); |
| RTC_DCHECK_NE(handshake_state_, kHandshakeWaitingForAck); |
| RTC_DCHECK_EQ(state_, kConnecting); |
| |
| if (config_.id == sid) { |
| return; |
| } |
| |
| const_cast<InternalDataChannelInit&>(config_).id = sid; |
| provider_->AddSctpDataStream(sid); |
| } |
| |
| void SctpDataChannel::OnClosingProcedureStartedRemotely(int sid) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| if (sid == config_.id && state_ != kClosing && state_ != kClosed) { |
| // Don't bother sending queued data since the side that initiated the |
| // closure wouldn't receive it anyway. See crbug.com/559394 for a lengthy |
| // discussion about this. |
| queued_send_data_.Clear(); |
| queued_control_data_.Clear(); |
| // Just need to change state to kClosing, SctpTransport will handle the |
| // rest of the closing procedure and OnClosingProcedureComplete will be |
| // called later. |
| started_closing_procedure_ = true; |
| SetState(kClosing); |
| } |
| } |
| |
| void SctpDataChannel::OnClosingProcedureComplete(int sid) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| if (sid == config_.id) { |
| // If the closing procedure is complete, we should have finished sending |
| // all pending data and transitioned to kClosing already. |
| RTC_DCHECK_EQ(state_, kClosing); |
| RTC_DCHECK(queued_send_data_.Empty()); |
| DisconnectFromProvider(); |
| SetState(kClosed); |
| } |
| } |
| |
| void SctpDataChannel::OnTransportChannelCreated() { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| if (!connected_to_provider_) { |
| connected_to_provider_ = provider_->ConnectDataChannel(this); |
| } |
| // The sid may have been unassigned when provider_->ConnectDataChannel was |
| // done. So always add the streams even if connected_to_provider_ is true. |
| if (config_.id >= 0) { |
| provider_->AddSctpDataStream(config_.id); |
| } |
| } |
| |
| void SctpDataChannel::OnTransportChannelClosed() { |
| // The SctpTransport is unusable (for example, because the SCTP m= section |
| // was rejected, or because the DTLS transport closed), so we need to close |
| // abruptly. |
| RTCError error = RTCError(RTCErrorType::OPERATION_ERROR_WITH_DATA, |
| "Transport channel closed"); |
| error.set_error_detail(RTCErrorDetailType::SCTP_FAILURE); |
| CloseAbruptlyWithError(std::move(error)); |
| } |
| |
| DataChannelStats SctpDataChannel::GetStats() const { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| DataChannelStats stats{internal_id_, id(), label(), |
| protocol(), state(), messages_sent(), |
| messages_received(), bytes_sent(), bytes_received()}; |
| return stats; |
| } |
| |
| void SctpDataChannel::OnDataReceived(const cricket::ReceiveDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| if (params.sid != config_.id) { |
| return; |
| } |
| |
| if (params.type == DataMessageType::kControl) { |
| if (handshake_state_ != kHandshakeWaitingForAck) { |
| // Ignore it if we are not expecting an ACK message. |
| RTC_LOG(LS_WARNING) |
| << "DataChannel received unexpected CONTROL message, sid = " |
| << params.sid; |
| return; |
| } |
| if (ParseDataChannelOpenAckMessage(payload)) { |
| // We can send unordered as soon as we receive the ACK message. |
| handshake_state_ = kHandshakeReady; |
| RTC_LOG(LS_INFO) << "DataChannel received OPEN_ACK message, sid = " |
| << params.sid; |
| } else { |
| RTC_LOG(LS_WARNING) |
| << "DataChannel failed to parse OPEN_ACK message, sid = " |
| << params.sid; |
| } |
| return; |
| } |
| |
| RTC_DCHECK(params.type == DataMessageType::kBinary || |
| params.type == DataMessageType::kText); |
| |
| RTC_LOG(LS_VERBOSE) << "DataChannel received DATA message, sid = " |
| << params.sid; |
| // We can send unordered as soon as we receive any DATA message since the |
| // remote side must have received the OPEN (and old clients do not send |
| // OPEN_ACK). |
| if (handshake_state_ == kHandshakeWaitingForAck) { |
| handshake_state_ = kHandshakeReady; |
| } |
| |
| bool binary = (params.type == webrtc::DataMessageType::kBinary); |
| auto buffer = std::make_unique<DataBuffer>(payload, binary); |
| if (state_ == kOpen && observer_) { |
| ++messages_received_; |
| bytes_received_ += buffer->size(); |
| observer_->OnMessage(*buffer.get()); |
| } else { |
| if (queued_received_data_.byte_count() + payload.size() > |
| kMaxQueuedReceivedDataBytes) { |
| RTC_LOG(LS_ERROR) << "Queued received data exceeds the max buffer size."; |
| |
| queued_received_data_.Clear(); |
| CloseAbruptlyWithError( |
| RTCError(RTCErrorType::RESOURCE_EXHAUSTED, |
| "Queued received data exceeds the max buffer size.")); |
| |
| return; |
| } |
| queued_received_data_.PushBack(std::move(buffer)); |
| } |
| } |
| |
| void SctpDataChannel::OnTransportReady(bool writable) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| |
| writable_ = writable; |
| if (!writable) { |
| return; |
| } |
| |
| SendQueuedControlMessages(); |
| SendQueuedDataMessages(); |
| |
| UpdateState(); |
| } |
| |
| void SctpDataChannel::CloseAbruptlyWithError(RTCError error) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| |
| if (state_ == kClosed) { |
| return; |
| } |
| |
| if (connected_to_provider_) { |
| DisconnectFromProvider(); |
| } |
| |
| // Closing abruptly means any queued data gets thrown away. |
| buffered_amount_ = 0; |
| |
| queued_send_data_.Clear(); |
| queued_control_data_.Clear(); |
| |
| // Still go to "kClosing" before "kClosed", since observers may be expecting |
| // that. |
| SetState(kClosing); |
| error_ = std::move(error); |
| SetState(kClosed); |
| } |
| |
| void SctpDataChannel::CloseAbruptlyWithDataChannelFailure( |
| const std::string& message) { |
| RTCError error(RTCErrorType::OPERATION_ERROR_WITH_DATA, message); |
| error.set_error_detail(RTCErrorDetailType::DATA_CHANNEL_FAILURE); |
| CloseAbruptlyWithError(std::move(error)); |
| } |
| |
| void SctpDataChannel::UpdateState() { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| // UpdateState determines what to do from a few state variables. Include |
| // all conditions required for each state transition here for |
| // clarity. OnTransportReady(true) will send any queued data and then invoke |
| // UpdateState(). |
| |
| switch (state_) { |
| case kConnecting: { |
| if (connected_to_provider_) { |
| if (handshake_state_ == kHandshakeShouldSendOpen) { |
| rtc::CopyOnWriteBuffer payload; |
| WriteDataChannelOpenMessage(label_, config_, &payload); |
| SendControlMessage(payload); |
| } else if (handshake_state_ == kHandshakeShouldSendAck) { |
| rtc::CopyOnWriteBuffer payload; |
| WriteDataChannelOpenAckMessage(&payload); |
| SendControlMessage(payload); |
| } |
| if (writable_ && (handshake_state_ == kHandshakeReady || |
| handshake_state_ == kHandshakeWaitingForAck)) { |
| SetState(kOpen); |
| // If we have received buffers before the channel got writable. |
| // Deliver them now. |
| DeliverQueuedReceivedData(); |
| } |
| } |
| break; |
| } |
| case kOpen: { |
| break; |
| } |
| case kClosing: { |
| // Wait for all queued data to be sent before beginning the closing |
| // procedure. |
| if (queued_send_data_.Empty() && queued_control_data_.Empty()) { |
| // For SCTP data channels, we need to wait for the closing procedure |
| // to complete; after calling RemoveSctpDataStream, |
| // OnClosingProcedureComplete will end up called asynchronously |
| // afterwards. |
| if (connected_to_provider_ && !started_closing_procedure_ && |
| config_.id >= 0) { |
| started_closing_procedure_ = true; |
| provider_->RemoveSctpDataStream(config_.id); |
| } |
| } |
| break; |
| } |
| case kClosed: |
| break; |
| } |
| } |
| |
| void SctpDataChannel::SetState(DataState state) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| if (state_ == state) { |
| return; |
| } |
| |
| state_ = state; |
| if (observer_) { |
| observer_->OnStateChange(); |
| } |
| if (state_ == kOpen) { |
| SignalOpened(this); |
| } else if (state_ == kClosed) { |
| SignalClosed(this); |
| } |
| } |
| |
| void SctpDataChannel::DisconnectFromProvider() { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| if (!connected_to_provider_) |
| return; |
| |
| provider_->DisconnectDataChannel(this); |
| connected_to_provider_ = false; |
| } |
| |
| void SctpDataChannel::DeliverQueuedReceivedData() { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| if (!observer_) { |
| return; |
| } |
| |
| while (!queued_received_data_.Empty()) { |
| std::unique_ptr<DataBuffer> buffer = queued_received_data_.PopFront(); |
| ++messages_received_; |
| bytes_received_ += buffer->size(); |
| observer_->OnMessage(*buffer); |
| } |
| } |
| |
| void SctpDataChannel::SendQueuedDataMessages() { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| if (queued_send_data_.Empty()) { |
| return; |
| } |
| |
| RTC_DCHECK(state_ == kOpen || state_ == kClosing); |
| |
| while (!queued_send_data_.Empty()) { |
| std::unique_ptr<DataBuffer> buffer = queued_send_data_.PopFront(); |
| if (!SendDataMessage(*buffer, false)) { |
| // Return the message to the front of the queue if sending is aborted. |
| queued_send_data_.PushFront(std::move(buffer)); |
| break; |
| } |
| } |
| } |
| |
| bool SctpDataChannel::SendDataMessage(const DataBuffer& buffer, |
| bool queue_if_blocked) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| SendDataParams send_params; |
| |
| send_params.ordered = config_.ordered; |
| // Send as ordered if it is still going through OPEN/ACK signaling. |
| if (handshake_state_ != kHandshakeReady && !config_.ordered) { |
| send_params.ordered = true; |
| RTC_LOG(LS_VERBOSE) |
| << "Sending data as ordered for unordered DataChannel " |
| "because the OPEN_ACK message has not been received."; |
| } |
| |
| send_params.max_rtx_count = config_.maxRetransmits; |
| send_params.max_rtx_ms = config_.maxRetransmitTime; |
| send_params.type = |
| buffer.binary ? DataMessageType::kBinary : DataMessageType::kText; |
| |
| cricket::SendDataResult send_result = cricket::SDR_SUCCESS; |
| bool success = |
| provider_->SendData(config_.id, send_params, buffer.data, &send_result); |
| |
| if (success) { |
| ++messages_sent_; |
| bytes_sent_ += buffer.size(); |
| |
| RTC_DCHECK(buffered_amount_ >= buffer.size()); |
| buffered_amount_ -= buffer.size(); |
| if (observer_ && buffer.size() > 0) { |
| observer_->OnBufferedAmountChange(buffer.size()); |
| } |
| return true; |
| } |
| |
| if (send_result == cricket::SDR_BLOCK) { |
| if (!queue_if_blocked || QueueSendDataMessage(buffer)) { |
| return false; |
| } |
| } |
| // Close the channel if the error is not SDR_BLOCK, or if queuing the |
| // message failed. |
| RTC_LOG(LS_ERROR) << "Closing the DataChannel due to a failure to send data, " |
| "send_result = " |
| << send_result; |
| CloseAbruptlyWithError( |
| RTCError(RTCErrorType::NETWORK_ERROR, "Failure to send data")); |
| |
| return false; |
| } |
| |
| bool SctpDataChannel::QueueSendDataMessage(const DataBuffer& buffer) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| size_t start_buffered_amount = queued_send_data_.byte_count(); |
| if (start_buffered_amount + buffer.size() > kMaxQueuedSendDataBytes) { |
| RTC_LOG(LS_ERROR) << "Can't buffer any more data for the data channel."; |
| return false; |
| } |
| queued_send_data_.PushBack(std::make_unique<DataBuffer>(buffer)); |
| return true; |
| } |
| |
| void SctpDataChannel::SendQueuedControlMessages() { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| PacketQueue control_packets; |
| control_packets.Swap(&queued_control_data_); |
| |
| while (!control_packets.Empty()) { |
| std::unique_ptr<DataBuffer> buf = control_packets.PopFront(); |
| SendControlMessage(buf->data); |
| } |
| } |
| |
| void SctpDataChannel::QueueControlMessage( |
| const rtc::CopyOnWriteBuffer& buffer) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| queued_control_data_.PushBack(std::make_unique<DataBuffer>(buffer, true)); |
| } |
| |
| bool SctpDataChannel::SendControlMessage(const rtc::CopyOnWriteBuffer& buffer) { |
| RTC_DCHECK_RUN_ON(signaling_thread_); |
| RTC_DCHECK(writable_); |
| RTC_DCHECK_GE(config_.id, 0); |
| |
| bool is_open_message = handshake_state_ == kHandshakeShouldSendOpen; |
| RTC_DCHECK(!is_open_message || !config_.negotiated); |
| |
| SendDataParams send_params; |
| // Send data as ordered before we receive any message from the remote peer to |
| // make sure the remote peer will not receive any data before it receives the |
| // OPEN message. |
| send_params.ordered = config_.ordered || is_open_message; |
| send_params.type = DataMessageType::kControl; |
| |
| cricket::SendDataResult send_result = cricket::SDR_SUCCESS; |
| bool retval = |
| provider_->SendData(config_.id, send_params, buffer, &send_result); |
| if (retval) { |
| RTC_LOG(LS_VERBOSE) << "Sent CONTROL message on channel " << config_.id; |
| |
| if (handshake_state_ == kHandshakeShouldSendAck) { |
| handshake_state_ = kHandshakeReady; |
| } else if (handshake_state_ == kHandshakeShouldSendOpen) { |
| handshake_state_ = kHandshakeWaitingForAck; |
| } |
| } else if (send_result == cricket::SDR_BLOCK) { |
| QueueControlMessage(buffer); |
| } else { |
| RTC_LOG(LS_ERROR) << "Closing the DataChannel due to a failure to send" |
| " the CONTROL message, send_result = " |
| << send_result; |
| CloseAbruptlyWithError(RTCError(RTCErrorType::NETWORK_ERROR, |
| "Failed to send a CONTROL message")); |
| } |
| return retval; |
| } |
| |
| // static |
| void SctpDataChannel::ResetInternalIdAllocatorForTesting(int new_value) { |
| g_unique_id = new_value; |
| } |
| |
| } // namespace webrtc |