blob: 359cc798c87118a9e59d0b767325127d1330ddb4 [file] [log] [blame]
/*
* 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