| /* |
| * 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/environment/environment.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/network/received_packet.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(const Environment& env, |
| rtc::Thread* network_thread, |
| rtc::PacketTransportInternal* transport); |
| DcSctpTransport(const Environment& env, |
| rtc::Thread* network_thread, |
| rtc::PacketTransportInternal* transport, |
| 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; |
| RTCError SendData(int sid, |
| const SendDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload) 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; |
| size_t buffered_amount(int sid) const override; |
| size_t buffered_amount_low_threshold(int sid) const override; |
| void SetBufferedAmountLowThreshold(int sid, size_t bytes) 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( |
| TaskQueueBase::DelayPrecision precision) override; |
| dcsctp::TimeMs TimeMillis() override; |
| uint32_t GetRandomInt(uint32_t low, uint32_t high) override; |
| void OnTotalBufferedAmountLow() override; |
| void OnBufferedAmountLow(dcsctp::StreamID stream_id) 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 rtc::ReceivedPacket& packet); |
| void MaybeConnectSocket(); |
| |
| rtc::Thread* network_thread_; |
| rtc::PacketTransportInternal* transport_; |
| Environment env_; |
| 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_ RTC_GUARDED_BY(network_thread_) = 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_ |