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* 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 <stdint.h>
#include <memory>
#include <set>
#include <string>
#include "absl/types/optional.h"
#include "api/data_channel_interface.h"
#include "api/priority.h"
#include "api/rtc_error.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/transport/data_channel_transport_interface.h"
#include "pc/data_channel_utils.h"
#include "pc/sctp_utils.h"
#include "rtc_base/containers/flat_set.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/ssl_stream_adapter.h" // For SSLRole
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/weak_ptr.h"
namespace webrtc {
class SctpDataChannel;
// Interface that acts as a bridge from the data channel to the transport.
// All methods in this interface need to be invoked on the network thread.
class SctpDataChannelControllerInterface {
// Sends the data to the transport.
virtual RTCError SendData(StreamId sid,
const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload) = 0;
// Adds the data channel SID to the transport for SCTP.
virtual void AddSctpDataStream(StreamId sid) = 0;
// Begins the closing procedure by sending an outgoing stream reset. Still
// need to wait for callbacks to tell when this completes.
virtual void RemoveSctpDataStream(StreamId sid) = 0;
// Notifies the controller of state changes.
virtual void OnChannelStateChanged(SctpDataChannel* data_channel,
DataChannelInterface::DataState state) = 0;
virtual size_t buffered_amount(StreamId sid) const = 0;
virtual size_t buffered_amount_low_threshold(StreamId sid) const = 0;
virtual void SetBufferedAmountLowThreshold(StreamId sid, size_t bytes) = 0;
virtual ~SctpDataChannelControllerInterface() {}
struct InternalDataChannelInit : public DataChannelInit {
enum OpenHandshakeRole { kOpener, kAcker, kNone };
// The default role is kOpener because the default `negotiated` is false.
InternalDataChannelInit() : open_handshake_role(kOpener) {}
explicit InternalDataChannelInit(const DataChannelInit& base);
// Does basic validation to determine if a data channel instance can be
// constructed using the configuration.
bool IsValid() const;
OpenHandshakeRole open_handshake_role;
// Optional fallback or backup flag from PC that's used for non-prenegotiated
// stream ids in situations where we cannot determine the SSL role from the
// transport for purposes of generating a stream ID.
// See:
absl::optional<rtc::SSLRole> fallback_ssl_role;
// Helper class to allocate unique IDs for SCTP DataChannels.
class SctpSidAllocator {
SctpSidAllocator() = default;
// Gets the first unused odd/even id based on the DTLS role. If `role` is
// SSL_CLIENT, the allocated id starts from 0 and takes even numbers;
// otherwise, the id starts from 1 and takes odd numbers.
// If a `StreamId` cannot be allocated, `absl::nullopt` is returned.
absl::optional<StreamId> AllocateSid(rtc::SSLRole role);
// Attempts to reserve a specific sid. Returns false if it's unavailable.
bool ReserveSid(StreamId sid);
// Indicates that `sid` isn't in use any more, and is thus available again.
void ReleaseSid(StreamId sid);
flat_set<StreamId> used_sids_ RTC_GUARDED_BY(&sequence_checker_);
RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_{
// SctpDataChannel is an implementation of the DataChannelInterface based on
// SctpTransport. It provides an implementation of unreliable or
// reliable data channels.
// DataChannel states:
// kConnecting: The channel has been created the transport might not yet be
// ready.
// kOpen: The open handshake has been performed (if relevant) and the data
// channel is able to send messages.
// kClosing: DataChannelInterface::Close has been called, or the remote side
// initiated the closing procedure, but the closing procedure has not
// yet finished.
// kClosed: The closing handshake is finished (possibly initiated from this,
// side, possibly from the peer).
// How the closing procedure works for SCTP:
// 1. Alice calls Close(), state changes to kClosing.
// 2. Alice finishes sending any queued data.
// 3. Alice calls RemoveSctpDataStream, sends outgoing stream reset.
// 4. Bob receives incoming stream reset; OnClosingProcedureStartedRemotely
// called.
// 5. Bob sends outgoing stream reset.
// 6. Alice receives incoming reset, Bob receives acknowledgement. Both receive
// OnClosingProcedureComplete callback and transition to kClosed.
class SctpDataChannel : public DataChannelInterface {
static rtc::scoped_refptr<SctpDataChannel> Create(
rtc::WeakPtr<SctpDataChannelControllerInterface> controller,
const std::string& label,
bool connected_to_transport,
const InternalDataChannelInit& config,
rtc::Thread* signaling_thread,
rtc::Thread* network_thread);
// Instantiates an API proxy for a SctpDataChannel instance that will be
// handed out to external callers.
// The `signaling_safety` flag is used for the ObserverAdapter callback proxy
// which delivers callbacks on the signaling thread but must not deliver such
// callbacks after the peerconnection has been closed. The data controller
// will update the flag when closed, which will cancel any pending event
// notifications.
static rtc::scoped_refptr<DataChannelInterface> CreateProxy(
rtc::scoped_refptr<SctpDataChannel> channel,
rtc::scoped_refptr<PendingTaskSafetyFlag> signaling_safety);
void RegisterObserver(DataChannelObserver* observer) override;
void UnregisterObserver() override;
std::string label() const override;
bool reliable() const override;
bool ordered() const override;
// Backwards compatible accessors
uint16_t maxRetransmitTime() const override;
uint16_t maxRetransmits() const override;
absl::optional<int> maxPacketLifeTime() const override;
absl::optional<int> maxRetransmitsOpt() const override;
std::string protocol() const override;
bool negotiated() const override;
int id() const override;
Priority priority() const override;
uint64_t buffered_amount() const override;
void Close() override;
DataState state() const override;
RTCError error() const override;
uint32_t messages_sent() const override;
uint64_t bytes_sent() const override;
uint32_t messages_received() const override;
uint64_t bytes_received() const override;
bool Send(const DataBuffer& buffer) override;
void SendAsync(DataBuffer buffer,
absl::AnyInvocable<void(RTCError) &&> on_complete) override;
// Close immediately, ignoring any queued data or closing procedure.
// This is called when the underlying SctpTransport is being destroyed.
// It is also called by the PeerConnection if SCTP ID assignment fails.
void CloseAbruptlyWithError(RTCError error);
// Specializations of CloseAbruptlyWithError
void CloseAbruptlyWithDataChannelFailure(const std::string& message);
// Called when the SctpTransport's ready to use. That can happen when we've
// finished negotiation, or if the channel was created after negotiation has
// already finished.
void OnTransportReady();
void OnDataReceived(DataMessageType type,
const rtc::CopyOnWriteBuffer& payload);
// Sets the SCTP sid and adds to transport layer if not set yet. Should only
// be called once.
void SetSctpSid_n(StreamId sid);
// The remote side started the closing procedure by resetting its outgoing
// stream (our incoming stream). Sets state to kClosing.
void OnClosingProcedureStartedRemotely();
// The closing procedure is complete; both incoming and outgoing stream
// resets are done and the channel can transition to kClosed. Called
// asynchronously after RemoveSctpDataStream.
void OnClosingProcedureComplete();
// Called when the transport channel is created.
void OnTransportChannelCreated();
// Called when the transport channel is unusable.
// This method makes sure the DataChannel is disconnected and changes state
// to kClosed.
void OnTransportChannelClosed(RTCError error);
// Called when the amount of data buffered to be sent falls to or below the
// threshold set when calling `SetBufferedAmountLowThreshold`.
void OnBufferedAmountLow();
DataChannelStats GetStats() const;
// Returns a unique identifier that's guaranteed to always be available,
// doesn't change throughout SctpDataChannel's lifetime and is used for
// stats purposes (see also `GetStats()`).
int internal_id() const { return internal_id_; }
absl::optional<StreamId> sid_n() const {
return id_n_;
// Reset the allocator for internal ID values for testing, so that
// the internal IDs generated are predictable. Test only.
static void ResetInternalIdAllocatorForTesting(int new_value);
SctpDataChannel(const InternalDataChannelInit& config,
rtc::WeakPtr<SctpDataChannelControllerInterface> controller,
const std::string& label,
bool connected_to_transport,
rtc::Thread* signaling_thread,
rtc::Thread* network_thread);
~SctpDataChannel() override;
class ObserverAdapter;
// The OPEN(_ACK) signaling state.
enum HandshakeState {
RTCError SendImpl(DataBuffer buffer) RTC_RUN_ON(network_thread_);
void UpdateState() RTC_RUN_ON(network_thread_);
void SetState(DataState state) RTC_RUN_ON(network_thread_);
void DeliverQueuedReceivedData() RTC_RUN_ON(network_thread_);
RTCError SendDataMessage(const DataBuffer& buffer, bool queue_if_blocked)
bool SendControlMessage(const rtc::CopyOnWriteBuffer& buffer)
bool connected_to_transport() const RTC_RUN_ON(network_thread_) {
return network_safety_->alive();
void MaybeSendOnBufferedAmountChanged() RTC_RUN_ON(network_thread_);
rtc::Thread* const signaling_thread_;
rtc::Thread* const network_thread_;
absl::optional<StreamId> id_n_ RTC_GUARDED_BY(network_thread_) =
const int internal_id_;
const std::string label_;
const std::string protocol_;
const absl::optional<int> max_retransmit_time_;
const absl::optional<int> max_retransmits_;
const absl::optional<Priority> priority_;
const bool negotiated_;
const bool ordered_;
// See the body of `MaybeSendOnBufferedAmountChanged`.
size_t expected_buffer_amount_ = 0;
DataChannelObserver* observer_ RTC_GUARDED_BY(network_thread_) = nullptr;
std::unique_ptr<ObserverAdapter> observer_adapter_;
DataState state_ RTC_GUARDED_BY(network_thread_) = kConnecting;
RTCError error_ RTC_GUARDED_BY(network_thread_);
uint32_t messages_sent_ RTC_GUARDED_BY(network_thread_) = 0;
uint64_t bytes_sent_ RTC_GUARDED_BY(network_thread_) = 0;
uint32_t messages_received_ RTC_GUARDED_BY(network_thread_) = 0;
uint64_t bytes_received_ RTC_GUARDED_BY(network_thread_) = 0;
rtc::WeakPtr<SctpDataChannelControllerInterface> controller_
HandshakeState handshake_state_ RTC_GUARDED_BY(network_thread_) =
// Did we already start the graceful SCTP closing procedure?
bool started_closing_procedure_ RTC_GUARDED_BY(network_thread_) = false;
PacketQueue queued_received_data_ RTC_GUARDED_BY(network_thread_);
rtc::scoped_refptr<PendingTaskSafetyFlag> network_safety_ =
} // namespace webrtc