/* * 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 #include #include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "api/array_view.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/types.h" #include "net/dcsctp/timer/task_queue_timeout.h" #include "p2p/base/packet_transport_internal.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 "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() override; // cricket::SctpTransportInternal 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 max_outbound_streams() const override; absl::optional max_inbound_streams() const override; void set_debug_name_for_testing(const char* debug_name) override; private: // dcsctp::DcSctpSocketCallbacks void SendPacket(rtc::ArrayView data) override; std::unique_ptr CreateTimeout() override; dcsctp::TimeMs TimeMillis() override; uint32_t GetRandomInt(uint32_t low, uint32_t high) override; void NotifyOutgoingMessageBufferEmpty() 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 outgoing_streams, absl::string_view reason) override; void OnStreamsResetPerformed( rtc::ArrayView outgoing_streams) override; void OnIncomingStreamsReset( rtc::ArrayView 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_; dcsctp::TaskQueueTimeoutFactory task_queue_timeout_factory_; std::unique_ptr socket_; std::string debug_name_ = "DcSctpTransport"; rtc::CopyOnWriteBuffer receive_buffer_; bool ready_to_send_data_ = false; }; } // namespace webrtc #endif // MEDIA_SCTP_DCSCTP_TRANSPORT_H_