| /* |
| * 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. |
| */ |
| |
| #ifndef PC_SCTP_DATA_CHANNEL_H_ |
| #define PC_SCTP_DATA_CHANNEL_H_ |
| |
| #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/transport/data_channel_transport_interface.h" |
| #include "media/base/media_channel.h" |
| #include "pc/data_channel_utils.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/ssl_stream_adapter.h" // For SSLRole |
| #include "rtc_base/third_party/sigslot/sigslot.h" |
| #include "rtc_base/thread.h" |
| #include "rtc_base/thread_annotations.h" |
| |
| namespace webrtc { |
| |
| class SctpDataChannel; |
| |
| // TODO(deadbeef): Get rid of this and have SctpDataChannel depend on |
| // SctpTransportInternal (pure virtual SctpTransport interface) instead. |
| class SctpDataChannelProviderInterface { |
| public: |
| // Sends the data to the transport. |
| virtual bool SendData(int sid, |
| const SendDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload, |
| cricket::SendDataResult* result) = 0; |
| // Connects to the transport signals. |
| virtual bool ConnectDataChannel(SctpDataChannel* data_channel) = 0; |
| // Disconnects from the transport signals. |
| virtual void DisconnectDataChannel(SctpDataChannel* data_channel) = 0; |
| // Adds the data channel SID to the transport for SCTP. |
| virtual void AddSctpDataStream(int 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(int sid) = 0; |
| // Returns true if the transport channel is ready to send data. |
| virtual bool ReadyToSendData() const = 0; |
| |
| protected: |
| virtual ~SctpDataChannelProviderInterface() {} |
| }; |
| |
| // TODO(tommi): Change to not inherit from DataChannelInit but to have it as |
| // a const member. Block access to the 'id' member since it cannot be const. |
| 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); |
| OpenHandshakeRole open_handshake_role; |
| }; |
| |
| // Helper class to allocate unique IDs for SCTP DataChannels. |
| class SctpSidAllocator { |
| public: |
| // 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. |
| // Returns false if no ID can be allocated. |
| bool AllocateSid(rtc::SSLRole role, int* sid); |
| |
| // Attempts to reserve a specific sid. Returns false if it's unavailable. |
| bool ReserveSid(int sid); |
| |
| // Indicates that |sid| isn't in use any more, and is thus available again. |
| void ReleaseSid(int sid); |
| |
| private: |
| // Checks if |sid| is available to be assigned to a new SCTP data channel. |
| bool IsSidAvailable(int sid) const; |
| |
| std::set<int> used_sids_; |
| }; |
| |
| // SctpDataChannel is an implementation of the DataChannelInterface based on |
| // SctpTransport. It provides an implementation of unreliable or |
| // reliabledata 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, |
| public sigslot::has_slots<> { |
| public: |
| static rtc::scoped_refptr<SctpDataChannel> Create( |
| SctpDataChannelProviderInterface* provider, |
| const std::string& label, |
| 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. |
| static rtc::scoped_refptr<DataChannelInterface> CreateProxy( |
| rtc::scoped_refptr<SctpDataChannel> channel); |
| |
| void RegisterObserver(DataChannelObserver* observer) override; |
| void UnregisterObserver() override; |
| |
| std::string label() const override { return label_; } |
| bool reliable() const override; |
| bool ordered() const override { return config_.ordered; } |
| // Backwards compatible accessors |
| uint16_t maxRetransmitTime() const override { |
| return config_.maxRetransmitTime ? *config_.maxRetransmitTime |
| : static_cast<uint16_t>(-1); |
| } |
| uint16_t maxRetransmits() const override { |
| return config_.maxRetransmits ? *config_.maxRetransmits |
| : static_cast<uint16_t>(-1); |
| } |
| absl::optional<int> maxPacketLifeTime() const override { |
| return config_.maxRetransmitTime; |
| } |
| absl::optional<int> maxRetransmitsOpt() const override { |
| return config_.maxRetransmits; |
| } |
| std::string protocol() const override { return config_.protocol; } |
| bool negotiated() const override { return config_.negotiated; } |
| int id() const override { return config_.id; } |
| Priority priority() const override { |
| return config_.priority ? *config_.priority : Priority::kLow; |
| } |
| |
| virtual int internal_id() const { return internal_id_; } |
| |
| 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; |
| |
| // 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); |
| void CloseAbruptlyWithSctpCauseCode(const std::string& message, |
| uint16_t cause_code); |
| |
| // Slots for provider to connect signals to. |
| // |
| // TODO(deadbeef): Make these private once we're hooking up signals ourselves, |
| // instead of relying on SctpDataChannelProviderInterface. |
| |
| // 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(bool writable); |
| |
| void OnDataReceived(const cricket::ReceiveDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload); |
| |
| // Sets the SCTP sid and adds to transport layer if not set yet. Should only |
| // be called once. |
| void SetSctpSid(int sid); |
| // The remote side started the closing procedure by resetting its outgoing |
| // stream (our incoming stream). Sets state to kClosing. |
| void OnClosingProcedureStartedRemotely(int sid); |
| // 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(int sid); |
| // Called when the transport channel is created. |
| // Only needs to be called for SCTP data channels. |
| void OnTransportChannelCreated(); |
| // Called when the transport channel is unusable. |
| // This method makes sure the DataChannel is disconnected and changes state |
| // to kClosed. |
| void OnTransportChannelClosed(); |
| |
| DataChannelStats GetStats() const; |
| |
| // Emitted when state transitions to kOpen. |
| sigslot::signal1<DataChannelInterface*> SignalOpened; |
| // Emitted when state transitions to kClosed. |
| // This signal can be used to tell when the channel's sid is free. |
| sigslot::signal1<DataChannelInterface*> SignalClosed; |
| |
| // 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); |
| |
| protected: |
| SctpDataChannel(const InternalDataChannelInit& config, |
| SctpDataChannelProviderInterface* client, |
| const std::string& label, |
| rtc::Thread* signaling_thread, |
| rtc::Thread* network_thread); |
| ~SctpDataChannel() override; |
| |
| private: |
| // The OPEN(_ACK) signaling state. |
| enum HandshakeState { |
| kHandshakeInit, |
| kHandshakeShouldSendOpen, |
| kHandshakeShouldSendAck, |
| kHandshakeWaitingForAck, |
| kHandshakeReady |
| }; |
| |
| bool Init(); |
| void UpdateState(); |
| void SetState(DataState state); |
| void DisconnectFromProvider(); |
| |
| void DeliverQueuedReceivedData(); |
| |
| void SendQueuedDataMessages(); |
| bool SendDataMessage(const DataBuffer& buffer, bool queue_if_blocked); |
| bool QueueSendDataMessage(const DataBuffer& buffer); |
| |
| void SendQueuedControlMessages(); |
| void QueueControlMessage(const rtc::CopyOnWriteBuffer& buffer); |
| bool SendControlMessage(const rtc::CopyOnWriteBuffer& buffer); |
| |
| rtc::Thread* const signaling_thread_; |
| rtc::Thread* const network_thread_; |
| const int internal_id_; |
| const std::string label_; |
| const InternalDataChannelInit config_; |
| DataChannelObserver* observer_ RTC_GUARDED_BY(signaling_thread_) = nullptr; |
| DataState state_ RTC_GUARDED_BY(signaling_thread_) = kConnecting; |
| RTCError error_ RTC_GUARDED_BY(signaling_thread_); |
| uint32_t messages_sent_ RTC_GUARDED_BY(signaling_thread_) = 0; |
| uint64_t bytes_sent_ RTC_GUARDED_BY(signaling_thread_) = 0; |
| uint32_t messages_received_ RTC_GUARDED_BY(signaling_thread_) = 0; |
| uint64_t bytes_received_ RTC_GUARDED_BY(signaling_thread_) = 0; |
| // Number of bytes of data that have been queued using Send(). Increased |
| // before each transport send and decreased after each successful send. |
| uint64_t buffered_amount_ RTC_GUARDED_BY(signaling_thread_) = 0; |
| SctpDataChannelProviderInterface* const provider_ |
| RTC_GUARDED_BY(signaling_thread_); |
| HandshakeState handshake_state_ RTC_GUARDED_BY(signaling_thread_) = |
| kHandshakeInit; |
| bool connected_to_provider_ RTC_GUARDED_BY(signaling_thread_) = false; |
| bool writable_ RTC_GUARDED_BY(signaling_thread_) = false; |
| // Did we already start the graceful SCTP closing procedure? |
| bool started_closing_procedure_ RTC_GUARDED_BY(signaling_thread_) = false; |
| // Control messages that always have to get sent out before any queued |
| // data. |
| PacketQueue queued_control_data_ RTC_GUARDED_BY(signaling_thread_); |
| PacketQueue queued_received_data_ RTC_GUARDED_BY(signaling_thread_); |
| PacketQueue queued_send_data_ RTC_GUARDED_BY(signaling_thread_); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // PC_SCTP_DATA_CHANNEL_H_ |