| /* |
| * 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/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 { |
| public: |
| // 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, PriorityValue priority) = 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; |
| |
| protected: |
| 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: https://www.rfc-editor.org/rfc/rfc8832.html#name-protocol-overview |
| absl::optional<rtc::SSLRole> fallback_ssl_role; |
| }; |
| |
| // Helper class to allocate unique IDs for SCTP DataChannels. |
| class SctpSidAllocator { |
| public: |
| 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); |
| |
| private: |
| flat_set<StreamId> used_sids_ RTC_GUARDED_BY(&sequence_checker_); |
| RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_{ |
| SequenceChecker::kDetached}; |
| }; |
| |
| // 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 { |
| public: |
| 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; |
| PriorityValue 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 { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| 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); |
| |
| protected: |
| 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; |
| |
| private: |
| class ObserverAdapter; |
| |
| // The OPEN(_ACK) signaling state. |
| enum HandshakeState { |
| kHandshakeInit, |
| kHandshakeShouldSendOpen, |
| kHandshakeShouldSendAck, |
| kHandshakeWaitingForAck, |
| kHandshakeReady |
| }; |
| |
| 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) |
| RTC_RUN_ON(network_thread_); |
| |
| bool SendControlMessage(const rtc::CopyOnWriteBuffer& buffer) |
| RTC_RUN_ON(network_thread_); |
| |
| 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_) = |
| absl::nullopt; |
| 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<PriorityValue> 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_ |
| RTC_GUARDED_BY(network_thread_); |
| HandshakeState handshake_state_ RTC_GUARDED_BY(network_thread_) = |
| kHandshakeInit; |
| // 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_ = |
| PendingTaskSafetyFlag::CreateDetachedInactive(); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // PC_SCTP_DATA_CHANNEL_H_ |