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/*
* Copyright (c) 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.
*/
#ifndef MEDIA_SCTP_DCSCTP_TRANSPORT_H_
#define MEDIA_SCTP_DCSCTP_TRANSPORT_H_
#include <memory>
#include <string>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/task_queue/task_queue_base.h"
#include "media/sctp/sctp_transport_internal.h"
#include "net/dcsctp/public/dcsctp_options.h"
#include "net/dcsctp/public/dcsctp_socket.h"
#include "net/dcsctp/public/dcsctp_socket_factory.h"
#include "net/dcsctp/public/types.h"
#include "net/dcsctp/timer/task_queue_timeout.h"
#include "p2p/base/packet_transport_internal.h"
#include "rtc_base/containers/flat_map.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/random.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
class DcSctpTransport : public cricket::SctpTransportInternal,
public dcsctp::DcSctpSocketCallbacks,
public sigslot::has_slots<> {
public:
DcSctpTransport(rtc::Thread* network_thread,
rtc::PacketTransportInternal* transport,
Clock* clock);
DcSctpTransport(rtc::Thread* network_thread,
rtc::PacketTransportInternal* transport,
Clock* clock,
std::unique_ptr<dcsctp::DcSctpSocketFactory> socket_factory);
~DcSctpTransport() override;
// cricket::SctpTransportInternal
void SetOnConnectedCallback(std::function<void()> callback) override;
void SetDataChannelSink(DataChannelSink* sink) override;
void SetDtlsTransport(rtc::PacketTransportInternal* transport) override;
bool Start(int local_sctp_port,
int remote_sctp_port,
int max_message_size) override;
bool OpenStream(int sid) override;
bool ResetStream(int sid) override;
bool SendData(int sid,
const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
cricket::SendDataResult* result = nullptr) override;
bool ReadyToSendData() override;
int max_message_size() const override;
absl::optional<int> max_outbound_streams() const override;
absl::optional<int> max_inbound_streams() const override;
void set_debug_name_for_testing(const char* debug_name) override;
private:
// dcsctp::DcSctpSocketCallbacks
dcsctp::SendPacketStatus SendPacketWithStatus(
rtc::ArrayView<const uint8_t> data) override;
std::unique_ptr<dcsctp::Timeout> CreateTimeout(
webrtc::TaskQueueBase::DelayPrecision precision) override;
dcsctp::TimeMs TimeMillis() override;
uint32_t GetRandomInt(uint32_t low, uint32_t high) override;
void OnTotalBufferedAmountLow() override;
void OnMessageReceived(dcsctp::DcSctpMessage message) override;
void OnError(dcsctp::ErrorKind error, absl::string_view message) override;
void OnAborted(dcsctp::ErrorKind error, absl::string_view message) override;
void OnConnected() override;
void OnClosed() override;
void OnConnectionRestarted() override;
void OnStreamsResetFailed(
rtc::ArrayView<const dcsctp::StreamID> outgoing_streams,
absl::string_view reason) override;
void OnStreamsResetPerformed(
rtc::ArrayView<const dcsctp::StreamID> outgoing_streams) override;
void OnIncomingStreamsReset(
rtc::ArrayView<const dcsctp::StreamID> incoming_streams) override;
// Transport callbacks
void ConnectTransportSignals();
void DisconnectTransportSignals();
void OnTransportWritableState(rtc::PacketTransportInternal* transport);
void OnTransportReadPacket(rtc::PacketTransportInternal* transport,
const char* data,
size_t length,
const int64_t& /* packet_time_us */,
int flags);
void OnTransportClosed(rtc::PacketTransportInternal* transport);
void MaybeConnectSocket();
rtc::Thread* network_thread_;
rtc::PacketTransportInternal* transport_;
Clock* clock_;
Random random_;
std::unique_ptr<dcsctp::DcSctpSocketFactory> socket_factory_;
dcsctp::TaskQueueTimeoutFactory task_queue_timeout_factory_;
std::unique_ptr<dcsctp::DcSctpSocketInterface> socket_;
std::string debug_name_ = "DcSctpTransport";
rtc::CopyOnWriteBuffer receive_buffer_;
// Used to keep track of the state of data channels.
// Reset needs to happen both ways before signaling the transport
// is closed.
struct StreamState {
// True when the local connection has initiated the reset.
// If a connection receives a reset for a stream that isn't
// already being reset locally, it needs to fire the signal
// SignalClosingProcedureStartedRemotely.
bool closure_initiated = false;
// True when the local connection received OnIncomingStreamsReset
bool incoming_reset_done = false;
// True when the local connection received OnStreamsResetPerformed
bool outgoing_reset_done = false;
};
// Map of all currently open or closing data channels
flat_map<dcsctp::StreamID, StreamState> stream_states_
RTC_GUARDED_BY(network_thread_);
bool ready_to_send_data_ = false;
std::function<void()> on_connected_callback_ RTC_GUARDED_BY(network_thread_);
DataChannelSink* data_channel_sink_ RTC_GUARDED_BY(network_thread_) = nullptr;
};
} // namespace webrtc
#endif // MEDIA_SCTP_DCSCTP_TRANSPORT_H_
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