Repo created

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/callback_deferrer.cc

@@ -0,0 +1,190 @@
+/*
+ *  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.
+ */
+#include "net/dcsctp/socket/callback_deferrer.h"
+
+#include "api/make_ref_counted.h"
+
+namespace dcsctp {
+
+void CallbackDeferrer::Prepare() {
+  RTC_DCHECK(!prepared_);
+  prepared_ = true;
+}
+
+void CallbackDeferrer::TriggerDeferred() {
+  // Need to swap here. The client may call into the library from within a
+  // callback, and that might result in adding new callbacks to this instance,
+  // and the vector can't be modified while iterated on.
+  RTC_DCHECK(prepared_);
+  prepared_ = false;
+  if (deferred_.empty()) {
+    return;
+  }
+  std::vector<std::pair<Callback, CallbackData>> deferred;
+  // Reserve a small buffer to prevent too much reallocation on growth.
+  deferred.reserve(8);
+  deferred.swap(deferred_);
+  for (auto& [cb, data] : deferred) {
+    cb(std::move(data), underlying_);
+  }
+}
+
+SendPacketStatus CallbackDeferrer::SendPacketWithStatus(
+    rtc::ArrayView<const uint8_t> data) {
+  // Will not be deferred - call directly.
+  return underlying_.SendPacketWithStatus(data);
+}
+
+std::unique_ptr<Timeout> CallbackDeferrer::CreateTimeout(
+    webrtc::TaskQueueBase::DelayPrecision precision) {
+  // Will not be deferred - call directly.
+  return underlying_.CreateTimeout(precision);
+}
+
+TimeMs CallbackDeferrer::TimeMillis() {
+  // This should not be called by the library - it's migrated to `Now()`.
+  RTC_DCHECK(false);
+  // Will not be deferred - call directly.
+  return underlying_.TimeMillis();
+}
+
+uint32_t CallbackDeferrer::GetRandomInt(uint32_t low, uint32_t high) {
+  // Will not be deferred - call directly.
+  return underlying_.GetRandomInt(low, high);
+}
+
+void CallbackDeferrer::OnMessageReceived(DcSctpMessage message) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      +[](CallbackData data, DcSctpSocketCallbacks& cb) {
+        return cb.OnMessageReceived(absl::get<DcSctpMessage>(std::move(data)));
+      },
+      std::move(message));
+}
+
+void CallbackDeferrer::OnError(ErrorKind error, absl::string_view message) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      +[](CallbackData data, DcSctpSocketCallbacks& cb) {
+        Error error = absl::get<Error>(std::move(data));
+        return cb.OnError(error.error, error.message);
+      },
+      Error{error, std::string(message)});
+}
+
+void CallbackDeferrer::OnAborted(ErrorKind error, absl::string_view message) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      +[](CallbackData data, DcSctpSocketCallbacks& cb) {
+        Error error = absl::get<Error>(std::move(data));
+        return cb.OnAborted(error.error, error.message);
+      },
+      Error{error, std::string(message)});
+}
+
+void CallbackDeferrer::OnConnected() {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      +[](CallbackData data, DcSctpSocketCallbacks& cb) {
+        return cb.OnConnected();
+      },
+      absl::monostate{});
+}
+
+void CallbackDeferrer::OnClosed() {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      +[](CallbackData data, DcSctpSocketCallbacks& cb) {
+        return cb.OnClosed();
+      },
+      absl::monostate{});
+}
+
+void CallbackDeferrer::OnConnectionRestarted() {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      +[](CallbackData data, DcSctpSocketCallbacks& cb) {
+        return cb.OnConnectionRestarted();
+      },
+      absl::monostate{});
+}
+
+void CallbackDeferrer::OnStreamsResetFailed(
+    rtc::ArrayView<const StreamID> outgoing_streams,
+    absl::string_view reason) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      +[](CallbackData data, DcSctpSocketCallbacks& cb) {
+        StreamReset stream_reset = absl::get<StreamReset>(std::move(data));
+        return cb.OnStreamsResetFailed(stream_reset.streams,
+                                       stream_reset.message);
+      },
+      StreamReset{{outgoing_streams.begin(), outgoing_streams.end()},
+                  std::string(reason)});
+}
+
+void CallbackDeferrer::OnStreamsResetPerformed(
+    rtc::ArrayView<const StreamID> outgoing_streams) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      +[](CallbackData data, DcSctpSocketCallbacks& cb) {
+        StreamReset stream_reset = absl::get<StreamReset>(std::move(data));
+        return cb.OnStreamsResetPerformed(stream_reset.streams);
+      },
+      StreamReset{{outgoing_streams.begin(), outgoing_streams.end()}});
+}
+
+void CallbackDeferrer::OnIncomingStreamsReset(
+    rtc::ArrayView<const StreamID> incoming_streams) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      +[](CallbackData data, DcSctpSocketCallbacks& cb) {
+        StreamReset stream_reset = absl::get<StreamReset>(std::move(data));
+        return cb.OnIncomingStreamsReset(stream_reset.streams);
+      },
+      StreamReset{{incoming_streams.begin(), incoming_streams.end()}});
+}
+
+void CallbackDeferrer::OnBufferedAmountLow(StreamID stream_id) {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      +[](CallbackData data, DcSctpSocketCallbacks& cb) {
+        return cb.OnBufferedAmountLow(absl::get<StreamID>(std::move(data)));
+      },
+      stream_id);
+}
+
+void CallbackDeferrer::OnTotalBufferedAmountLow() {
+  RTC_DCHECK(prepared_);
+  deferred_.emplace_back(
+      +[](CallbackData data, DcSctpSocketCallbacks& cb) {
+        return cb.OnTotalBufferedAmountLow();
+      },
+      absl::monostate{});
+}
+
+void CallbackDeferrer::OnLifecycleMessageExpired(LifecycleId lifecycle_id,
+                                                 bool maybe_delivered) {
+  // Will not be deferred - call directly.
+  underlying_.OnLifecycleMessageExpired(lifecycle_id, maybe_delivered);
+}
+void CallbackDeferrer::OnLifecycleMessageFullySent(LifecycleId lifecycle_id) {
+  // Will not be deferred - call directly.
+  underlying_.OnLifecycleMessageFullySent(lifecycle_id);
+}
+void CallbackDeferrer::OnLifecycleMessageDelivered(LifecycleId lifecycle_id) {
+  // Will not be deferred - call directly.
+  underlying_.OnLifecycleMessageDelivered(lifecycle_id);
+}
+void CallbackDeferrer::OnLifecycleEnd(LifecycleId lifecycle_id) {
+  // Will not be deferred - call directly.
+  underlying_.OnLifecycleEnd(lifecycle_id);
+}
+}  // namespace dcsctp

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/callback_deferrer.h

@@ -0,0 +1,116 @@
+/*
+ *  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 NET_DCSCTP_SOCKET_CALLBACK_DEFERRER_H_
+#define NET_DCSCTP_SOCKET_CALLBACK_DEFERRER_H_
+
+#include <cstdint>
+#include <functional>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/variant.h"
+#include "api/array_view.h"
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+
+namespace dcsctp {
+// Defers callbacks until they can be safely triggered.
+//
+// There are a lot of callbacks from the dcSCTP library to the client,
+// such as when messages are received or streams are closed. When the client
+// receives these callbacks, the client is expected to be able to call into the
+// library - from within the callback. For example, sending a reply message when
+// a certain SCTP message has been received, or to reconnect when the connection
+// was closed for any reason. This means that the dcSCTP library must always be
+// in a consistent and stable state when these callbacks are delivered, and to
+// ensure that's the case, callbacks are not immediately delivered from where
+// they originate, but instead queued (deferred) by this class. At the end of
+// any public API method that may result in callbacks, they are triggered and
+// then delivered.
+//
+// There are a number of exceptions, which is clearly annotated in the API.
+class CallbackDeferrer : public DcSctpSocketCallbacks {
+ public:
+  class ScopedDeferrer {
+   public:
+    explicit ScopedDeferrer(CallbackDeferrer& callback_deferrer)
+        : callback_deferrer_(callback_deferrer) {
+      callback_deferrer_.Prepare();
+    }
+
+    ~ScopedDeferrer() { callback_deferrer_.TriggerDeferred(); }
+
+   private:
+    CallbackDeferrer& callback_deferrer_;
+  };
+
+  explicit CallbackDeferrer(DcSctpSocketCallbacks& underlying)
+      : underlying_(underlying) {}
+
+  // Implementation of DcSctpSocketCallbacks
+  SendPacketStatus SendPacketWithStatus(
+      rtc::ArrayView<const uint8_t> data) override;
+  std::unique_ptr<Timeout> CreateTimeout(
+      webrtc::TaskQueueBase::DelayPrecision precision) override;
+  TimeMs TimeMillis() override;
+  webrtc::Timestamp Now() override { return underlying_.Now(); }
+  uint32_t GetRandomInt(uint32_t low, uint32_t high) override;
+  void OnMessageReceived(DcSctpMessage message) override;
+  void OnError(ErrorKind error, absl::string_view message) override;
+  void OnAborted(ErrorKind error, absl::string_view message) override;
+  void OnConnected() override;
+  void OnClosed() override;
+  void OnConnectionRestarted() override;
+  void OnStreamsResetFailed(rtc::ArrayView<const StreamID> outgoing_streams,
+                            absl::string_view reason) override;
+  void OnStreamsResetPerformed(
+      rtc::ArrayView<const StreamID> outgoing_streams) override;
+  void OnIncomingStreamsReset(
+      rtc::ArrayView<const StreamID> incoming_streams) override;
+  void OnBufferedAmountLow(StreamID stream_id) override;
+  void OnTotalBufferedAmountLow() override;
+
+  void OnLifecycleMessageExpired(LifecycleId lifecycle_id,
+                                 bool maybe_delivered) override;
+  void OnLifecycleMessageFullySent(LifecycleId lifecycle_id) override;
+  void OnLifecycleMessageDelivered(LifecycleId lifecycle_id) override;
+  void OnLifecycleEnd(LifecycleId lifecycle_id) override;
+
+ private:
+  struct Error {
+    ErrorKind error;
+    std::string message;
+  };
+  struct StreamReset {
+    std::vector<StreamID> streams;
+    std::string message;
+  };
+  // Use a pre-sized variant for storage to avoid double heap allocation. This
+  // variant can hold all cases of stored data.
+  using CallbackData = absl::
+      variant<absl::monostate, DcSctpMessage, Error, StreamReset, StreamID>;
+  using Callback = void (*)(CallbackData, DcSctpSocketCallbacks&);
+
+  void Prepare();
+  void TriggerDeferred();
+
+  DcSctpSocketCallbacks& underlying_;
+  bool prepared_ = false;
+  std::vector<std::pair<Callback, CallbackData>> deferred_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_CALLBACK_DEFERRER_H_

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/capabilities.h

@@ -0,0 +1,32 @@
+/*
+ *  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 NET_DCSCTP_SOCKET_CAPABILITIES_H_
+#define NET_DCSCTP_SOCKET_CAPABILITIES_H_
+
+#include <cstdint>
+namespace dcsctp {
+// Indicates what the association supports, meaning that both parties
+// support it and that feature can be used.
+struct Capabilities {
+  // RFC3758 Partial Reliability Extension
+  bool partial_reliability = false;
+  // RFC8260 Stream Schedulers and User Message Interleaving
+  bool message_interleaving = false;
+  // RFC6525 Stream Reconfiguration
+  bool reconfig = false;
+  // https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-zero-checksum/
+  bool zero_checksum = false;
+  // Negotiated maximum incoming and outgoing stream count.
+  uint16_t negotiated_maximum_incoming_streams = 0;
+  uint16_t negotiated_maximum_outgoing_streams = 0;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_CAPABILITIES_H_

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/context.h

@@ -0,0 +1,67 @@
+/*
+ *  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 NET_DCSCTP_SOCKET_CONTEXT_H_
+#define NET_DCSCTP_SOCKET_CONTEXT_H_
+
+#include <cstdint>
+
+#include "absl/strings/string_view.h"
+#include "api/units/time_delta.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+// A set of helper methods used by handlers to e.g. send packets.
+//
+// Implemented by the TransmissionControlBlock.
+class Context {
+ public:
+  virtual ~Context() = default;
+
+  // Indicates if a connection has been established.
+  virtual bool is_connection_established() const = 0;
+
+  // Returns this side's initial TSN value.
+  virtual TSN my_initial_tsn() const = 0;
+
+  // Returns the peer's initial TSN value.
+  virtual TSN peer_initial_tsn() const = 0;
+
+  // Returns the socket callbacks.
+  virtual DcSctpSocketCallbacks& callbacks() const = 0;
+
+  // Observes a measured RTT value.
+  virtual void ObserveRTT(webrtc::TimeDelta rtt_ms) = 0;
+
+  // Returns the current Retransmission Timeout (rto) value, in milliseconds.
+  virtual webrtc::TimeDelta current_rto() const = 0;
+
+  // Increments the transmission error counter, given a human readable reason.
+  virtual bool IncrementTxErrorCounter(absl::string_view reason) = 0;
+
+  // Clears the transmission error counter.
+  virtual void ClearTxErrorCounter() = 0;
+
+  // Returns true if there have been too many retransmission errors.
+  virtual bool HasTooManyTxErrors() const = 0;
+
+  // Returns a PacketBuilder, filled in with the correct verification tag.
+  virtual SctpPacket::Builder PacketBuilder() const = 0;
+
+  // Builds the packet from `builder` and sends it.
+  virtual void Send(SctpPacket::Builder& builder) = 0;
+};
+
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_CONTEXT_H_

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/dcsctp_socket.h

@@ -0,0 +1,301 @@
+/*
+ *  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 NET_DCSCTP_SOCKET_DCSCTP_SOCKET_H_
+#define NET_DCSCTP_SOCKET_DCSCTP_SOCKET_H_
+
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/chunk/abort_chunk.h"
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/chunk/cookie_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/cookie_echo_chunk.h"
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+#include "net/dcsctp/packet/chunk/data_common.h"
+#include "net/dcsctp/packet/chunk/error_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_common.h"
+#include "net/dcsctp/packet/chunk/heartbeat_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/heartbeat_request_chunk.h"
+#include "net/dcsctp/packet/chunk/idata_chunk.h"
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/init_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/init_chunk.h"
+#include "net/dcsctp/packet/chunk/reconfig_chunk.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/packet/chunk/shutdown_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/shutdown_chunk.h"
+#include "net/dcsctp/packet/chunk/shutdown_complete_chunk.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/packet_observer.h"
+#include "net/dcsctp/rx/data_tracker.h"
+#include "net/dcsctp/rx/reassembly_queue.h"
+#include "net/dcsctp/socket/callback_deferrer.h"
+#include "net/dcsctp/socket/packet_sender.h"
+#include "net/dcsctp/socket/state_cookie.h"
+#include "net/dcsctp/socket/transmission_control_block.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/retransmission_error_counter.h"
+#include "net/dcsctp/tx/retransmission_queue.h"
+#include "net/dcsctp/tx/retransmission_timeout.h"
+#include "net/dcsctp/tx/rr_send_queue.h"
+
+namespace dcsctp {
+
+// DcSctpSocket represents a single SCTP socket, to be used over DTLS.
+//
+// Every dcSCTP is completely isolated from any other socket.
+//
+// This class manages all packet and chunk dispatching and mainly handles the
+// connection sequences (connect, close, shutdown, etc) as well as managing
+// the Transmission Control Block (tcb).
+//
+// This class is thread-compatible.
+class DcSctpSocket : public DcSctpSocketInterface {
+ public:
+  // Instantiates a DcSctpSocket, which interacts with the world through the
+  // `callbacks` interface and is configured using `options`.
+  //
+  // For debugging, `log_prefix` will prefix all debug logs, and a
+  // `packet_observer` can be attached to e.g. dump sent and received packets.
+  DcSctpSocket(absl::string_view log_prefix,
+               DcSctpSocketCallbacks& callbacks,
+               std::unique_ptr<PacketObserver> packet_observer,
+               const DcSctpOptions& options);
+
+  DcSctpSocket(const DcSctpSocket&) = delete;
+  DcSctpSocket& operator=(const DcSctpSocket&) = delete;
+
+  // Implementation of `DcSctpSocketInterface`.
+  void ReceivePacket(rtc::ArrayView<const uint8_t> data) override;
+  void HandleTimeout(TimeoutID timeout_id) override;
+  void Connect() override;
+  void RestoreFromState(const DcSctpSocketHandoverState& state) override;
+  void Shutdown() override;
+  void Close() override;
+  SendStatus Send(DcSctpMessage message,
+                  const SendOptions& send_options) override;
+  std::vector<SendStatus> SendMany(rtc::ArrayView<DcSctpMessage> messages,
+                                   const SendOptions& send_options) override;
+  ResetStreamsStatus ResetStreams(
+      rtc::ArrayView<const StreamID> outgoing_streams) override;
+  SocketState state() const override;
+  const DcSctpOptions& options() const override { return options_; }
+  void SetMaxMessageSize(size_t max_message_size) override;
+  void SetStreamPriority(StreamID stream_id, StreamPriority priority) override;
+  StreamPriority GetStreamPriority(StreamID stream_id) const override;
+  size_t buffered_amount(StreamID stream_id) const override;
+  size_t buffered_amount_low_threshold(StreamID stream_id) const override;
+  void SetBufferedAmountLowThreshold(StreamID stream_id, size_t bytes) override;
+  absl::optional<Metrics> GetMetrics() const override;
+  HandoverReadinessStatus GetHandoverReadiness() const override;
+  absl::optional<DcSctpSocketHandoverState> GetHandoverStateAndClose() override;
+  SctpImplementation peer_implementation() const override {
+    return metrics_.peer_implementation;
+  }
+  // Returns this socket's verification tag, or zero if not yet connected.
+  VerificationTag verification_tag() const {
+    return tcb_ != nullptr ? tcb_->my_verification_tag() : VerificationTag(0);
+  }
+
+ private:
+  // Parameter proposals valid during the connect phase.
+  struct ConnectParameters {
+    TSN initial_tsn = TSN(0);
+    VerificationTag verification_tag = VerificationTag(0);
+  };
+
+  // Detailed state (separate from SocketState, which is the public state).
+  enum class State {
+    kClosed,
+    kCookieWait,
+    // TCB valid in these:
+    kCookieEchoed,
+    kEstablished,
+    kShutdownPending,
+    kShutdownSent,
+    kShutdownReceived,
+    kShutdownAckSent,
+  };
+
+  // Returns the log prefix used for debug logging.
+  std::string log_prefix() const;
+
+  bool IsConsistent() const;
+  static constexpr absl::string_view ToString(DcSctpSocket::State state);
+
+  void CreateTransmissionControlBlock(const Capabilities& capabilities,
+                                      VerificationTag my_verification_tag,
+                                      TSN my_initial_tsn,
+                                      VerificationTag peer_verification_tag,
+                                      TSN peer_initial_tsn,
+                                      size_t a_rwnd,
+                                      TieTag tie_tag);
+
+  // Changes the socket state, given a `reason` (for debugging/logging).
+  void SetState(State state, absl::string_view reason);
+  // Closes the association. Note that the TCB will not be valid past this call.
+  void InternalClose(ErrorKind error, absl::string_view message);
+  // Closes the association, because of too many retransmission errors.
+  void CloseConnectionBecauseOfTooManyTransmissionErrors();
+  // Timer expiration handlers
+  webrtc::TimeDelta OnInitTimerExpiry();
+  webrtc::TimeDelta OnCookieTimerExpiry();
+  webrtc::TimeDelta OnShutdownTimerExpiry();
+  void OnSentPacket(rtc::ArrayView<const uint8_t> packet,
+                    SendPacketStatus status);
+  // Sends SHUTDOWN or SHUTDOWN-ACK if the socket is shutting down and if all
+  // outstanding data has been acknowledged.
+  void MaybeSendShutdownOrAck();
+  // If the socket is shutting down, responds SHUTDOWN to any incoming DATA.
+  void MaybeSendShutdownOnPacketReceived(const SctpPacket& packet);
+  // If there are streams pending to be reset, send a request to reset them.
+  void MaybeSendResetStreamsRequest();
+  // Performs internal processing shared between Send and SendMany.
+  SendStatus InternalSend(const DcSctpMessage& message,
+                          const SendOptions& send_options);
+  // Sends a INIT chunk.
+  void SendInit();
+  // Sends a SHUTDOWN chunk.
+  void SendShutdown();
+  // Sends a SHUTDOWN-ACK chunk.
+  void SendShutdownAck();
+  // Validates the SCTP packet, as a whole - not the validity of individual
+  // chunks within it, as that's done in the different chunk handlers.
+  bool ValidatePacket(const SctpPacket& packet);
+  // Parses `payload`, which is a serialized packet that is just going to be
+  // sent and prints all chunks.
+  void DebugPrintOutgoing(rtc::ArrayView<const uint8_t> payload);
+  // Called whenever data has been received, or the cumulative acknowledgment
+  // TSN has moved, that may result in delivering messages.
+  void MaybeDeliverMessages();
+  // Returns true if there is a TCB, and false otherwise (and reports an error).
+  bool ValidateHasTCB();
+
+  // Returns true if the parsing of a chunk of type `T` succeeded. If it didn't,
+  // it reports an error and returns false.
+  template <class T>
+  bool ValidateParseSuccess(const absl::optional<T>& c) {
+    if (c.has_value()) {
+      return true;
+    }
+
+    ReportFailedToParseChunk(T::kType);
+    return false;
+  }
+
+  // Reports failing to have parsed a chunk with the provided `chunk_type`.
+  void ReportFailedToParseChunk(int chunk_type);
+  // Called when unknown chunks are received. May report an error.
+  bool HandleUnrecognizedChunk(const SctpPacket::ChunkDescriptor& descriptor);
+
+  // Will dispatch more specific chunk handlers.
+  bool Dispatch(const CommonHeader& header,
+                const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming DATA chunks.
+  void HandleData(const CommonHeader& header,
+                  const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming I-DATA chunks.
+  void HandleIData(const CommonHeader& header,
+                   const SctpPacket::ChunkDescriptor& descriptor);
+  // Common handler for DATA and I-DATA chunks.
+  void HandleDataCommon(AnyDataChunk& chunk);
+  // Handles incoming INIT chunks.
+  void HandleInit(const CommonHeader& header,
+                  const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming INIT-ACK chunks.
+  void HandleInitAck(const CommonHeader& header,
+                     const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming SACK chunks.
+  void HandleSack(const CommonHeader& header,
+                  const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming HEARTBEAT chunks.
+  void HandleHeartbeatRequest(const CommonHeader& header,
+                              const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming HEARTBEAT-ACK chunks.
+  void HandleHeartbeatAck(const CommonHeader& header,
+                          const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming ABORT chunks.
+  void HandleAbort(const CommonHeader& header,
+                   const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming ERROR chunks.
+  void HandleError(const CommonHeader& header,
+                   const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming COOKIE-ECHO chunks.
+  void HandleCookieEcho(const CommonHeader& header,
+                        const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles receiving COOKIE-ECHO when there already is a TCB. The return value
+  // indicates if the processing should continue.
+  bool HandleCookieEchoWithTCB(const CommonHeader& header,
+                               const StateCookie& cookie);
+  // Handles incoming COOKIE-ACK chunks.
+  void HandleCookieAck(const CommonHeader& header,
+                       const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming SHUTDOWN chunks.
+  void HandleShutdown(const CommonHeader& header,
+                      const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming SHUTDOWN-ACK chunks.
+  void HandleShutdownAck(const CommonHeader& header,
+                         const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming FORWARD-TSN chunks.
+  void HandleForwardTsn(const CommonHeader& header,
+                        const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming I-FORWARD-TSN chunks.
+  void HandleIForwardTsn(const CommonHeader& header,
+                         const SctpPacket::ChunkDescriptor& descriptor);
+  // Handles incoming RE-CONFIG chunks.
+  void HandleReconfig(const CommonHeader& header,
+                      const SctpPacket::ChunkDescriptor& descriptor);
+  // Common handled for FORWARD-TSN/I-FORWARD-TSN.
+  void HandleForwardTsnCommon(const AnyForwardTsnChunk& chunk);
+  // Handles incoming SHUTDOWN-COMPLETE chunks
+  void HandleShutdownComplete(const CommonHeader& header,
+                              const SctpPacket::ChunkDescriptor& descriptor);
+
+  const std::string log_prefix_;
+  const std::unique_ptr<PacketObserver> packet_observer_;
+  Metrics metrics_;
+  DcSctpOptions options_;
+
+  // Enqueues callbacks and dispatches them just before returning to the caller.
+  CallbackDeferrer callbacks_;
+
+  TimerManager timer_manager_;
+  const std::unique_ptr<Timer> t1_init_;
+  const std::unique_ptr<Timer> t1_cookie_;
+  const std::unique_ptr<Timer> t2_shutdown_;
+
+  // Packets that failed to be sent, but should be retried.
+  PacketSender packet_sender_;
+
+  // The actual SendQueue implementation. As data can be sent on a socket before
+  // the connection is established, this component is not in the TCB.
+  RRSendQueue send_queue_;
+
+  // Contains verification tag and initial TSN between having sent the INIT
+  // until the connection is established (there is no TCB at this point).
+  ConnectParameters connect_params_;
+  // The socket state.
+  State state_ = State::kClosed;
+  // If the connection is established, contains a transmission control block.
+  std::unique_ptr<TransmissionControlBlock> tcb_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_DCSCTP_SOCKET_H_

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/heartbeat_handler.cc

@@ -0,0 +1,201 @@
+/*
+ *  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.
+ */
+#include "net/dcsctp/socket/heartbeat_handler.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/bind_front.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/units/time_delta.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/packet/chunk/heartbeat_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/heartbeat_request_chunk.h"
+#include "net/dcsctp/packet/parameter/heartbeat_info_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/socket/context.h"
+#include "net/dcsctp/timer/timer.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+using ::webrtc::TimeDelta;
+using ::webrtc::Timestamp;
+
+// This is stored (in serialized form) as HeartbeatInfoParameter sent in
+// HeartbeatRequestChunk and received back in HeartbeatAckChunk. It should be
+// well understood that this data may be modified by the peer, so it can't
+// be trusted.
+//
+// It currently only stores a timestamp, in millisecond precision, to allow for
+// RTT measurements. If that would be manipulated by the peer, it would just
+// result in incorrect RTT measurements, which isn't an issue.
+class HeartbeatInfo {
+ public:
+  static constexpr size_t kBufferSize = sizeof(uint64_t);
+  static_assert(kBufferSize == 8, "Unexpected buffer size");
+
+  explicit HeartbeatInfo(Timestamp created_at) : created_at_(created_at) {}
+
+  std::vector<uint8_t> Serialize() {
+    uint32_t high_bits = static_cast<uint32_t>(created_at_.ms() >> 32);
+    uint32_t low_bits = static_cast<uint32_t>(created_at_.ms());
+
+    std::vector<uint8_t> data(kBufferSize);
+    BoundedByteWriter<kBufferSize> writer(data);
+    writer.Store32<0>(high_bits);
+    writer.Store32<4>(low_bits);
+    return data;
+  }
+
+  static absl::optional<HeartbeatInfo> Deserialize(
+      rtc::ArrayView<const uint8_t> data) {
+    if (data.size() != kBufferSize) {
+      RTC_LOG(LS_WARNING) << "Invalid heartbeat info: " << data.size()
+                          << " bytes";
+      return absl::nullopt;
+    }
+
+    BoundedByteReader<kBufferSize> reader(data);
+    uint32_t high_bits = reader.Load32<0>();
+    uint32_t low_bits = reader.Load32<4>();
+
+    uint64_t created_at = static_cast<uint64_t>(high_bits) << 32 | low_bits;
+    return HeartbeatInfo(Timestamp::Millis(created_at));
+  }
+
+  Timestamp created_at() const { return created_at_; }
+
+ private:
+  const Timestamp created_at_;
+};
+
+HeartbeatHandler::HeartbeatHandler(absl::string_view log_prefix,
+                                   const DcSctpOptions& options,
+                                   Context* context,
+                                   TimerManager* timer_manager)
+    : log_prefix_(log_prefix),
+      ctx_(context),
+      timer_manager_(timer_manager),
+      interval_duration_(options.heartbeat_interval.ToTimeDelta()),
+      interval_duration_should_include_rtt_(
+          options.heartbeat_interval_include_rtt),
+      interval_timer_(timer_manager_->CreateTimer(
+          "heartbeat-interval",
+          absl::bind_front(&HeartbeatHandler::OnIntervalTimerExpiry, this),
+          TimerOptions(interval_duration_,
+                       TimerBackoffAlgorithm::kFixed))),
+      timeout_timer_(timer_manager_->CreateTimer(
+          "heartbeat-timeout",
+          absl::bind_front(&HeartbeatHandler::OnTimeoutTimerExpiry, this),
+          TimerOptions(options.rto_initial.ToTimeDelta(),
+                       TimerBackoffAlgorithm::kExponential,
+                       /*max_restarts=*/0))) {
+  // The interval timer must always be running as long as the association is up.
+  RestartTimer();
+}
+
+void HeartbeatHandler::RestartTimer() {
+  if (interval_duration_.IsZero()) {
+    // Heartbeating has been disabled.
+    return;
+  }
+
+  if (interval_duration_should_include_rtt_) {
+    // The RTT should be used, but it's not easy accessible. The RTO will
+    // suffice.
+    interval_timer_->set_duration(
+        interval_duration_ + ctx_->current_rto());
+  } else {
+    interval_timer_->set_duration(interval_duration_);
+  }
+
+  interval_timer_->Start();
+}
+
+void HeartbeatHandler::HandleHeartbeatRequest(HeartbeatRequestChunk chunk) {
+  // https://tools.ietf.org/html/rfc4960#section-8.3
+  // "The receiver of the HEARTBEAT should immediately respond with a
+  // HEARTBEAT ACK that contains the Heartbeat Information TLV, together with
+  // any other received TLVs, copied unchanged from the received HEARTBEAT
+  // chunk."
+  ctx_->Send(ctx_->PacketBuilder().Add(
+      HeartbeatAckChunk(std::move(chunk).extract_parameters())));
+}
+
+void HeartbeatHandler::HandleHeartbeatAck(HeartbeatAckChunk chunk) {
+  timeout_timer_->Stop();
+  absl::optional<HeartbeatInfoParameter> info_param = chunk.info();
+  if (!info_param.has_value()) {
+    ctx_->callbacks().OnError(
+        ErrorKind::kParseFailed,
+        "Failed to parse HEARTBEAT-ACK; No Heartbeat Info parameter");
+    return;
+  }
+  absl::optional<HeartbeatInfo> info =
+      HeartbeatInfo::Deserialize(info_param->info());
+  if (!info.has_value()) {
+    ctx_->callbacks().OnError(ErrorKind::kParseFailed,
+                              "Failed to parse HEARTBEAT-ACK; Failed to "
+                              "deserialized Heartbeat info parameter");
+    return;
+  }
+
+  Timestamp now = ctx_->callbacks().Now();
+  if (info->created_at() > Timestamp::Zero() && info->created_at() <= now) {
+    ctx_->ObserveRTT(now - info->created_at());
+  }
+
+  // https://tools.ietf.org/html/rfc4960#section-8.1
+  // "The counter shall be reset each time ... a HEARTBEAT ACK is received from
+  // the peer endpoint."
+  ctx_->ClearTxErrorCounter();
+}
+
+TimeDelta HeartbeatHandler::OnIntervalTimerExpiry() {
+  if (ctx_->is_connection_established()) {
+    HeartbeatInfo info(ctx_->callbacks().Now());
+    timeout_timer_->set_duration(ctx_->current_rto());
+    timeout_timer_->Start();
+    RTC_DLOG(LS_INFO) << log_prefix_ << "Sending HEARTBEAT with timeout "
+                      << webrtc::ToString(timeout_timer_->duration());
+
+    Parameters parameters = Parameters::Builder()
+                                .Add(HeartbeatInfoParameter(info.Serialize()))
+                                .Build();
+
+    ctx_->Send(ctx_->PacketBuilder().Add(
+        HeartbeatRequestChunk(std::move(parameters))));
+  } else {
+    RTC_DLOG(LS_VERBOSE)
+        << log_prefix_
+        << "Will not send HEARTBEAT when connection not established";
+  }
+  return TimeDelta::Zero();
+}
+
+TimeDelta HeartbeatHandler::OnTimeoutTimerExpiry() {
+  // Note that the timeout timer is not restarted. It will be started again when
+  // the interval timer expires.
+  RTC_DCHECK(!timeout_timer_->is_running());
+  ctx_->IncrementTxErrorCounter("HEARTBEAT timeout");
+  return TimeDelta::Zero();
+}
+}  // namespace dcsctp

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/heartbeat_handler.h

@@ -0,0 +1,69 @@
+/*
+ *  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 NET_DCSCTP_SOCKET_HEARTBEAT_HANDLER_H_
+#define NET_DCSCTP_SOCKET_HEARTBEAT_HANDLER_H_
+
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "net/dcsctp/packet/chunk/heartbeat_ack_chunk.h"
+#include "net/dcsctp/packet/chunk/heartbeat_request_chunk.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/socket/context.h"
+#include "net/dcsctp/timer/timer.h"
+
+namespace dcsctp {
+
+// HeartbeatHandler handles all logic around sending heartbeats and receiving
+// the responses, as well as receiving incoming heartbeat requests.
+//
+// Heartbeats are sent on idle connections to ensure that the connection is
+// still healthy and to measure the RTT. If a number of heartbeats time out,
+// the connection will eventually be closed.
+class HeartbeatHandler {
+ public:
+  HeartbeatHandler(absl::string_view log_prefix,
+                   const DcSctpOptions& options,
+                   Context* context,
+                   TimerManager* timer_manager);
+
+  // Called when the heartbeat interval timer should be restarted. This is
+  // generally done every time data is sent, which makes the timer expire when
+  // the connection is idle.
+  void RestartTimer();
+
+  // Called on received HeartbeatRequestChunk chunks.
+  void HandleHeartbeatRequest(HeartbeatRequestChunk chunk);
+
+  // Called on received HeartbeatRequestChunk chunks.
+  void HandleHeartbeatAck(HeartbeatAckChunk chunk);
+
+ private:
+  webrtc::TimeDelta OnIntervalTimerExpiry();
+  webrtc::TimeDelta OnTimeoutTimerExpiry();
+
+  const absl::string_view log_prefix_;
+  Context* ctx_;
+  TimerManager* timer_manager_;
+  // The time for a connection to be idle before a heartbeat is sent.
+  const webrtc::TimeDelta interval_duration_;
+  // Adding RTT to the duration will add some jitter, which is good in
+  // production, but less good in unit tests, which is why it can be disabled.
+  const bool interval_duration_should_include_rtt_;
+  const std::unique_ptr<Timer> interval_timer_;
+  const std::unique_ptr<Timer> timeout_timer_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_HEARTBEAT_HANDLER_H_

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/mock_context.h

@@ -0,0 +1,73 @@
+/*
+ *  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 NET_DCSCTP_SOCKET_MOCK_CONTEXT_H_
+#define NET_DCSCTP_SOCKET_MOCK_CONTEXT_H_
+
+#include <cstdint>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/socket/context.h"
+#include "net/dcsctp/socket/mock_dcsctp_socket_callbacks.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+
+class MockContext : public Context {
+ public:
+  static constexpr TSN MyInitialTsn() { return TSN(990); }
+  static constexpr TSN PeerInitialTsn() { return TSN(10); }
+  static constexpr VerificationTag PeerVerificationTag() {
+    return VerificationTag(0x01234567);
+  }
+
+  explicit MockContext(MockDcSctpSocketCallbacks* callbacks)
+      : callbacks_(*callbacks) {
+    ON_CALL(*this, is_connection_established)
+        .WillByDefault(testing::Return(true));
+    ON_CALL(*this, my_initial_tsn)
+        .WillByDefault(testing::Return(MyInitialTsn()));
+    ON_CALL(*this, peer_initial_tsn)
+        .WillByDefault(testing::Return(PeerInitialTsn()));
+    ON_CALL(*this, callbacks).WillByDefault(testing::ReturnRef(callbacks_));
+    ON_CALL(*this, current_rto)
+        .WillByDefault(testing::Return(webrtc::TimeDelta::Millis(123)));
+    ON_CALL(*this, Send).WillByDefault([this](SctpPacket::Builder& builder) {
+      callbacks_.SendPacketWithStatus(builder.Build());
+    });
+  }
+
+  MOCK_METHOD(bool, is_connection_established, (), (const, override));
+  MOCK_METHOD(TSN, my_initial_tsn, (), (const, override));
+  MOCK_METHOD(TSN, peer_initial_tsn, (), (const, override));
+  MOCK_METHOD(DcSctpSocketCallbacks&, callbacks, (), (const, override));
+
+  MOCK_METHOD(void, ObserveRTT, (webrtc::TimeDelta rtt), (override));
+  MOCK_METHOD(webrtc::TimeDelta, current_rto, (), (const, override));
+  MOCK_METHOD(bool,
+              IncrementTxErrorCounter,
+              (absl::string_view reason),
+              (override));
+  MOCK_METHOD(void, ClearTxErrorCounter, (), (override));
+  MOCK_METHOD(bool, HasTooManyTxErrors, (), (const, override));
+  SctpPacket::Builder PacketBuilder() const override {
+    return SctpPacket::Builder(PeerVerificationTag(), options_);
+  }
+  MOCK_METHOD(void, Send, (SctpPacket::Builder & builder), (override));
+
+  DcSctpOptions options_;
+  MockDcSctpSocketCallbacks& callbacks_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_MOCK_CONTEXT_H_

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/mock_dcsctp_socket_callbacks.h

@@ -0,0 +1,183 @@
+/*
+ *  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 NET_DCSCTP_SOCKET_MOCK_DCSCTP_SOCKET_CALLBACKS_H_
+#define NET_DCSCTP_SOCKET_MOCK_DCSCTP_SOCKET_CALLBACKS_H_
+
+#include <cstdint>
+#include <deque>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#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 "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/public/timeout.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/timer/fake_timeout.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/random.h"
+#include "test/gmock.h"
+
+namespace dcsctp {
+
+namespace internal {
+// It can be argued if a mocked random number generator should be deterministic
+// or if it should be have as a "real" random number generator. In this
+// implementation, each instantiation of `MockDcSctpSocketCallbacks` will have
+// their `GetRandomInt` return different sequences, but each instantiation will
+// always generate the same sequence of random numbers. This to make it easier
+// to compare logs from tests, but still to let e.g. two different sockets (used
+// in the same test) get different random numbers, so that they don't start e.g.
+// on the same sequence number. While that isn't an issue in the protocol, it
+// just makes debugging harder as the two sockets would look exactly the same.
+//
+// In a real implementation of `DcSctpSocketCallbacks` the random number
+// generator backing `GetRandomInt` should be seeded externally and correctly.
+inline int GetUniqueSeed() {
+  static int seed = 0;
+  return ++seed;
+}
+}  // namespace internal
+
+class MockDcSctpSocketCallbacks : public DcSctpSocketCallbacks {
+ public:
+  explicit MockDcSctpSocketCallbacks(absl::string_view name = "")
+      : log_prefix_(name.empty() ? "" : std::string(name) + ": "),
+        random_(internal::GetUniqueSeed()),
+        timeout_manager_([this]() { return now_; }) {
+    ON_CALL(*this, SendPacketWithStatus)
+        .WillByDefault([this](rtc::ArrayView<const uint8_t> data) {
+          sent_packets_.emplace_back(
+              std::vector<uint8_t>(data.begin(), data.end()));
+          return SendPacketStatus::kSuccess;
+        });
+    ON_CALL(*this, OnMessageReceived)
+        .WillByDefault([this](DcSctpMessage message) {
+          received_messages_.emplace_back(std::move(message));
+        });
+
+    ON_CALL(*this, OnError)
+        .WillByDefault([this](ErrorKind error, absl::string_view message) {
+          RTC_LOG(LS_WARNING)
+              << log_prefix_ << "Socket error: " << ToString(error) << "; "
+              << message;
+        });
+    ON_CALL(*this, OnAborted)
+        .WillByDefault([this](ErrorKind error, absl::string_view message) {
+          RTC_LOG(LS_WARNING)
+              << log_prefix_ << "Socket abort: " << ToString(error) << "; "
+              << message;
+        });
+    ON_CALL(*this, Now).WillByDefault([this]() { return now_; });
+  }
+
+  MOCK_METHOD(SendPacketStatus,
+              SendPacketWithStatus,
+              (rtc::ArrayView<const uint8_t> data),
+              (override));
+
+  std::unique_ptr<Timeout> CreateTimeout(
+      webrtc::TaskQueueBase::DelayPrecision precision) override {
+    // The fake timeout manager does not implement |precision|.
+    return timeout_manager_.CreateTimeout();
+  }
+
+  MOCK_METHOD(webrtc::Timestamp, Now, (), (override));
+  uint32_t GetRandomInt(uint32_t low, uint32_t high) override {
+    return random_.Rand(low, high);
+  }
+
+  MOCK_METHOD(void, OnMessageReceived, (DcSctpMessage message), (override));
+  MOCK_METHOD(void,
+              OnError,
+              (ErrorKind error, absl::string_view message),
+              (override));
+  MOCK_METHOD(void,
+              OnAborted,
+              (ErrorKind error, absl::string_view message),
+              (override));
+  MOCK_METHOD(void, OnConnected, (), (override));
+  MOCK_METHOD(void, OnClosed, (), (override));
+  MOCK_METHOD(void, OnConnectionRestarted, (), (override));
+  MOCK_METHOD(void,
+              OnStreamsResetFailed,
+              (rtc::ArrayView<const StreamID> outgoing_streams,
+               absl::string_view reason),
+              (override));
+  MOCK_METHOD(void,
+              OnStreamsResetPerformed,
+              (rtc::ArrayView<const StreamID> outgoing_streams),
+              (override));
+  MOCK_METHOD(void,
+              OnIncomingStreamsReset,
+              (rtc::ArrayView<const StreamID> incoming_streams),
+              (override));
+  MOCK_METHOD(void, OnBufferedAmountLow, (StreamID stream_id), (override));
+  MOCK_METHOD(void, OnTotalBufferedAmountLow, (), (override));
+  MOCK_METHOD(void,
+              OnLifecycleMessageExpired,
+              (LifecycleId lifecycle_id, bool maybe_delivered),
+              (override));
+  MOCK_METHOD(void,
+              OnLifecycleMessageFullySent,
+              (LifecycleId lifecycle_id),
+              (override));
+  MOCK_METHOD(void,
+              OnLifecycleMessageDelivered,
+              (LifecycleId lifecycle_id),
+              (override));
+  MOCK_METHOD(void, OnLifecycleEnd, (LifecycleId lifecycle_id), (override));
+
+  bool HasPacket() const { return !sent_packets_.empty(); }
+
+  std::vector<uint8_t> ConsumeSentPacket() {
+    if (sent_packets_.empty()) {
+      return {};
+    }
+    std::vector<uint8_t> ret = std::move(sent_packets_.front());
+    sent_packets_.pop_front();
+    return ret;
+  }
+  absl::optional<DcSctpMessage> ConsumeReceivedMessage() {
+    if (received_messages_.empty()) {
+      return absl::nullopt;
+    }
+    DcSctpMessage ret = std::move(received_messages_.front());
+    received_messages_.pop_front();
+    return ret;
+  }
+
+  void AdvanceTime(webrtc::TimeDelta duration) { now_ = now_ + duration; }
+  void SetTime(webrtc::Timestamp now) { now_ = now; }
+
+  absl::optional<TimeoutID> GetNextExpiredTimeout() {
+    return timeout_manager_.GetNextExpiredTimeout();
+  }
+
+  webrtc::TimeDelta GetTimeToNextTimeout() const {
+    return timeout_manager_.GetTimeToNextTimeout();
+  }
+
+ private:
+  const std::string log_prefix_;
+  webrtc::Timestamp now_ = webrtc::Timestamp::Zero();
+  webrtc::Random random_;
+  FakeTimeoutManager timeout_manager_;
+  std::deque<std::vector<uint8_t>> sent_packets_;
+  std::deque<DcSctpMessage> received_messages_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_MOCK_DCSCTP_SOCKET_CALLBACKS_H_

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/packet_sender.cc

@@ -0,0 +1,48 @@
+/*
+ *  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.
+ */
+#include "net/dcsctp/socket/packet_sender.h"
+
+#include <utility>
+#include <vector>
+
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+PacketSender::PacketSender(DcSctpSocketCallbacks& callbacks,
+                           std::function<void(rtc::ArrayView<const uint8_t>,
+                                              SendPacketStatus)> on_sent_packet)
+    : callbacks_(callbacks), on_sent_packet_(std::move(on_sent_packet)) {}
+
+bool PacketSender::Send(SctpPacket::Builder& builder, bool write_checksum) {
+  if (builder.empty()) {
+    return false;
+  }
+
+  std::vector<uint8_t> payload = builder.Build(write_checksum);
+
+  SendPacketStatus status = callbacks_.SendPacketWithStatus(payload);
+  on_sent_packet_(payload, status);
+  switch (status) {
+    case SendPacketStatus::kSuccess: {
+      return true;
+    }
+    case SendPacketStatus::kTemporaryFailure: {
+      // TODO(boivie): Queue this packet to be retried to be sent later.
+      return false;
+    }
+
+    case SendPacketStatus::kError: {
+      // Nothing that can be done.
+      return false;
+    }
+  }
+}
+}  // namespace dcsctp

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/packet_sender.h

@@ -0,0 +1,40 @@
+/*
+ *  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 NET_DCSCTP_SOCKET_PACKET_SENDER_H_
+#define NET_DCSCTP_SOCKET_PACKET_SENDER_H_
+
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+
+namespace dcsctp {
+
+// The PacketSender sends packets to the network using the provided callback
+// interface. When an attempt to send a packet is made, the `on_sent_packet`
+// callback will be triggered.
+class PacketSender {
+ public:
+  PacketSender(DcSctpSocketCallbacks& callbacks,
+               std::function<void(rtc::ArrayView<const uint8_t>,
+                                  SendPacketStatus)> on_sent_packet);
+
+  // Sends the packet, and returns true if it was sent successfully.
+  bool Send(SctpPacket::Builder& builder, bool write_checksum = true);
+
+ private:
+  DcSctpSocketCallbacks& callbacks_;
+
+  // Callback that will be triggered for every send attempt, indicating the
+  // status of the operation.
+  std::function<void(rtc::ArrayView<const uint8_t>, SendPacketStatus)>
+      on_sent_packet_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_PACKET_SENDER_H_

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/state_cookie.cc

@@ -0,0 +1,88 @@
+/*
+ *  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.
+ */
+#include "net/dcsctp/socket/state_cookie.h"
+
+#include <cstdint>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/packet/bounded_byte_reader.h"
+#include "net/dcsctp/packet/bounded_byte_writer.h"
+#include "net/dcsctp/socket/capabilities.h"
+#include "rtc_base/logging.h"
+
+namespace dcsctp {
+
+// Magic values, which the state cookie is prefixed with.
+constexpr uint32_t kMagic1 = 1684230979;
+constexpr uint32_t kMagic2 = 1414541360;
+constexpr size_t StateCookie::kCookieSize;
+
+std::vector<uint8_t> StateCookie::Serialize() {
+  std::vector<uint8_t> cookie;
+  cookie.resize(kCookieSize);
+  BoundedByteWriter<kCookieSize> buffer(cookie);
+  buffer.Store32<0>(kMagic1);
+  buffer.Store32<4>(kMagic2);
+  buffer.Store32<8>(*peer_tag_);
+  buffer.Store32<12>(*my_tag_);
+  buffer.Store32<16>(*peer_initial_tsn_);
+  buffer.Store32<20>(*my_initial_tsn_);
+  buffer.Store32<24>(a_rwnd_);
+  buffer.Store32<28>(static_cast<uint32_t>(*tie_tag_ >> 32));
+  buffer.Store32<32>(static_cast<uint32_t>(*tie_tag_));
+  buffer.Store8<36>(capabilities_.partial_reliability);
+  buffer.Store8<37>(capabilities_.message_interleaving);
+  buffer.Store8<38>(capabilities_.reconfig);
+  buffer.Store16<40>(capabilities_.negotiated_maximum_incoming_streams);
+  buffer.Store16<42>(capabilities_.negotiated_maximum_outgoing_streams);
+  buffer.Store8<44>(capabilities_.zero_checksum);
+  return cookie;
+}
+
+absl::optional<StateCookie> StateCookie::Deserialize(
+    rtc::ArrayView<const uint8_t> cookie) {
+  if (cookie.size() != kCookieSize) {
+    RTC_DLOG(LS_WARNING) << "Invalid state cookie: " << cookie.size()
+                         << " bytes";
+    return absl::nullopt;
+  }
+
+  BoundedByteReader<kCookieSize> buffer(cookie);
+  uint32_t magic1 = buffer.Load32<0>();
+  uint32_t magic2 = buffer.Load32<4>();
+  if (magic1 != kMagic1 || magic2 != kMagic2) {
+    RTC_DLOG(LS_WARNING) << "Invalid state cookie; wrong magic";
+    return absl::nullopt;
+  }
+
+  VerificationTag peer_tag(buffer.Load32<8>());
+  VerificationTag my_tag(buffer.Load32<12>());
+  TSN peer_initial_tsn(buffer.Load32<16>());
+  TSN my_initial_tsn(buffer.Load32<20>());
+  uint32_t a_rwnd = buffer.Load32<24>();
+  uint32_t tie_tag_upper = buffer.Load32<28>();
+  uint32_t tie_tag_lower = buffer.Load32<32>();
+  TieTag tie_tag(static_cast<uint64_t>(tie_tag_upper) << 32 |
+                 static_cast<uint64_t>(tie_tag_lower));
+  Capabilities capabilities;
+  capabilities.partial_reliability = buffer.Load8<36>() != 0;
+  capabilities.message_interleaving = buffer.Load8<37>() != 0;
+  capabilities.reconfig = buffer.Load8<38>() != 0;
+  capabilities.negotiated_maximum_incoming_streams = buffer.Load16<40>();
+  capabilities.negotiated_maximum_outgoing_streams = buffer.Load16<42>();
+  capabilities.zero_checksum = buffer.Load8<44>() != 0;
+
+  return StateCookie(peer_tag, my_tag, peer_initial_tsn, my_initial_tsn, a_rwnd,
+                     tie_tag, capabilities);
+}
+
+}  // namespace dcsctp

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/state_cookie.h

@@ -0,0 +1,75 @@
+/*
+ *  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 NET_DCSCTP_SOCKET_STATE_COOKIE_H_
+#define NET_DCSCTP_SOCKET_STATE_COOKIE_H_
+
+#include <cstdint>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/socket/capabilities.h"
+
+namespace dcsctp {
+
+// This is serialized as a state cookie and put in INIT_ACK. The client then
+// responds with this in COOKIE_ECHO.
+//
+// NOTE: Expect that the client will modify it to try to exploit the library.
+// Do not trust anything in it; no pointers or anything like that.
+class StateCookie {
+ public:
+  static constexpr size_t kCookieSize = 45;
+
+  StateCookie(VerificationTag peer_tag,
+              VerificationTag my_tag,
+              TSN peer_initial_tsn,
+              TSN my_initial_tsn,
+              uint32_t a_rwnd,
+              TieTag tie_tag,
+              Capabilities capabilities)
+      : peer_tag_(peer_tag),
+        my_tag_(my_tag),
+        peer_initial_tsn_(peer_initial_tsn),
+        my_initial_tsn_(my_initial_tsn),
+        a_rwnd_(a_rwnd),
+        tie_tag_(tie_tag),
+        capabilities_(capabilities) {}
+
+  // Returns a serialized version of this cookie.
+  std::vector<uint8_t> Serialize();
+
+  // Deserializes the cookie, and returns absl::nullopt if that failed.
+  static absl::optional<StateCookie> Deserialize(
+      rtc::ArrayView<const uint8_t> cookie);
+
+  VerificationTag peer_tag() const { return peer_tag_; }
+  VerificationTag my_tag() const { return my_tag_; }
+  TSN peer_initial_tsn() const { return peer_initial_tsn_; }
+  TSN my_initial_tsn() const { return my_initial_tsn_; }
+  uint32_t a_rwnd() const { return a_rwnd_; }
+  TieTag tie_tag() const { return tie_tag_; }
+  const Capabilities& capabilities() const { return capabilities_; }
+
+ private:
+  // Also called "Tag_A" in RFC4960.
+  const VerificationTag peer_tag_;
+  // Also called "Tag_Z" in RFC4960.
+  const VerificationTag my_tag_;
+  const TSN peer_initial_tsn_;
+  const TSN my_initial_tsn_;
+  const uint32_t a_rwnd_;
+  const TieTag tie_tag_;
+  const Capabilities capabilities_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_STATE_COOKIE_H_

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/stream_reset_handler.cc

@@ -0,0 +1,387 @@
+/*
+ *  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.
+ */
+#include "net/dcsctp/socket/stream_reset_handler.h"
+
+#include <cstdint>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/units/time_delta.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/chunk/reconfig_chunk.h"
+#include "net/dcsctp/packet/parameter/add_incoming_streams_request_parameter.h"
+#include "net/dcsctp/packet/parameter/add_outgoing_streams_request_parameter.h"
+#include "net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/parameter.h"
+#include "net/dcsctp/packet/parameter/reconfiguration_response_parameter.h"
+#include "net/dcsctp/packet/parameter/ssn_tsn_reset_request_parameter.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/packet/tlv_trait.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/rx/data_tracker.h"
+#include "net/dcsctp/rx/reassembly_queue.h"
+#include "net/dcsctp/socket/context.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/retransmission_queue.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/str_join.h"
+
+namespace dcsctp {
+namespace {
+using ::webrtc::TimeDelta;
+using ResponseResult = ReconfigurationResponseParameter::Result;
+
+bool DescriptorsAre(const std::vector<ParameterDescriptor>& c,
+                    uint16_t e1,
+                    uint16_t e2) {
+  return (c[0].type == e1 && c[1].type == e2) ||
+         (c[0].type == e2 && c[1].type == e1);
+}
+
+}  // namespace
+
+bool StreamResetHandler::Validate(const ReConfigChunk& chunk) {
+  const Parameters& parameters = chunk.parameters();
+
+  // https://tools.ietf.org/html/rfc6525#section-3.1
+  // "Note that each RE-CONFIG chunk holds at least one parameter
+  // and at most two parameters. Only the following combinations are allowed:"
+  std::vector<ParameterDescriptor> descriptors = parameters.descriptors();
+  if (descriptors.size() == 1) {
+    if ((descriptors[0].type == OutgoingSSNResetRequestParameter::kType) ||
+        (descriptors[0].type == IncomingSSNResetRequestParameter::kType) ||
+        (descriptors[0].type == SSNTSNResetRequestParameter::kType) ||
+        (descriptors[0].type == AddOutgoingStreamsRequestParameter::kType) ||
+        (descriptors[0].type == AddIncomingStreamsRequestParameter::kType) ||
+        (descriptors[0].type == ReconfigurationResponseParameter::kType)) {
+      return true;
+    }
+  } else if (descriptors.size() == 2) {
+    if (DescriptorsAre(descriptors, OutgoingSSNResetRequestParameter::kType,
+                       IncomingSSNResetRequestParameter::kType) ||
+        DescriptorsAre(descriptors, AddOutgoingStreamsRequestParameter::kType,
+                       AddIncomingStreamsRequestParameter::kType) ||
+        DescriptorsAre(descriptors, ReconfigurationResponseParameter::kType,
+                       OutgoingSSNResetRequestParameter::kType) ||
+        DescriptorsAre(descriptors, ReconfigurationResponseParameter::kType,
+                       ReconfigurationResponseParameter::kType)) {
+      return true;
+    }
+  }
+
+  RTC_LOG(LS_WARNING) << "Invalid set of RE-CONFIG parameters";
+  return false;
+}
+
+absl::optional<std::vector<ReconfigurationResponseParameter>>
+StreamResetHandler::Process(const ReConfigChunk& chunk) {
+  if (!Validate(chunk)) {
+    return absl::nullopt;
+  }
+
+  std::vector<ReconfigurationResponseParameter> responses;
+
+  for (const ParameterDescriptor& desc : chunk.parameters().descriptors()) {
+    switch (desc.type) {
+      case OutgoingSSNResetRequestParameter::kType:
+        HandleResetOutgoing(desc, responses);
+        break;
+
+      case IncomingSSNResetRequestParameter::kType:
+        HandleResetIncoming(desc, responses);
+        break;
+
+      case ReconfigurationResponseParameter::kType:
+        HandleResponse(desc);
+        break;
+    }
+  }
+
+  return responses;
+}
+
+void StreamResetHandler::HandleReConfig(ReConfigChunk chunk) {
+  absl::optional<std::vector<ReconfigurationResponseParameter>> responses =
+      Process(chunk);
+
+  if (!responses.has_value()) {
+    ctx_->callbacks().OnError(ErrorKind::kParseFailed,
+                              "Failed to parse RE-CONFIG command");
+    return;
+  }
+
+  if (!responses->empty()) {
+    SctpPacket::Builder b = ctx_->PacketBuilder();
+    Parameters::Builder params_builder;
+    for (const auto& response : *responses) {
+      params_builder.Add(response);
+    }
+    b.Add(ReConfigChunk(params_builder.Build()));
+    ctx_->Send(b);
+  }
+}
+
+bool StreamResetHandler::ValidateReqSeqNbr(
+    UnwrappedReconfigRequestSn req_seq_nbr,
+    std::vector<ReconfigurationResponseParameter>& responses) {
+  if (req_seq_nbr == last_processed_req_seq_nbr_) {
+    // https://www.rfc-editor.org/rfc/rfc6525.html#section-5.2.1 "If the
+    // received RE-CONFIG chunk contains at least one request and based on the
+    // analysis of the Re-configuration Request Sequence Numbers this is the
+    // last received RE-CONFIG chunk (i.e., a retransmission), the same
+    // RE-CONFIG chunk MUST to be sent back in response, as it was earlier."
+    RTC_DLOG(LS_VERBOSE) << log_prefix_ << "req=" << *req_seq_nbr
+                         << " already processed, returning result="
+                         << ToString(last_processed_req_result_);
+    responses.push_back(ReconfigurationResponseParameter(
+        req_seq_nbr.Wrap(), last_processed_req_result_));
+    return false;
+  }
+
+  if (req_seq_nbr != last_processed_req_seq_nbr_.next_value()) {
+    // Too old, too new, from wrong association etc.
+    // This is expected to happen when handing over a RTCPeerConnection from one
+    // server to another. The client will notice this and may decide to close
+    // old data channels, which may be sent to the wrong (or both) servers
+    // during a handover.
+    RTC_DLOG(LS_VERBOSE) << log_prefix_ << "req=" << *req_seq_nbr
+                         << " bad seq_nbr";
+    responses.push_back(ReconfigurationResponseParameter(
+        req_seq_nbr.Wrap(), ResponseResult::kErrorBadSequenceNumber));
+    return false;
+  }
+
+  return true;
+}
+
+void StreamResetHandler::HandleResetOutgoing(
+    const ParameterDescriptor& descriptor,
+    std::vector<ReconfigurationResponseParameter>& responses) {
+  absl::optional<OutgoingSSNResetRequestParameter> req =
+      OutgoingSSNResetRequestParameter::Parse(descriptor.data);
+  if (!req.has_value()) {
+    ctx_->callbacks().OnError(ErrorKind::kParseFailed,
+                              "Failed to parse Outgoing Reset command");
+    return;
+  }
+
+  UnwrappedReconfigRequestSn request_sn =
+      incoming_reconfig_request_sn_unwrapper_.Unwrap(
+          req->request_sequence_number());
+
+  if (ValidateReqSeqNbr(request_sn, responses)) {
+    last_processed_req_seq_nbr_ = request_sn;
+    if (data_tracker_->IsLaterThanCumulativeAckedTsn(
+            req->sender_last_assigned_tsn())) {
+      // https://datatracker.ietf.org/doc/html/rfc6525#section-5.2.2
+      // E2) "If the Sender's Last Assigned TSN is greater than the cumulative
+      // acknowledgment point, then the endpoint MUST enter 'deferred reset
+      // processing'."
+      reassembly_queue_->EnterDeferredReset(req->sender_last_assigned_tsn(),
+                                            req->stream_ids());
+      // "If the endpoint enters 'deferred reset processing', it MUST put a
+      // Re-configuration Response Parameter into a RE-CONFIG chunk indicating
+      // 'In progress' and MUST send the RE-CONFIG chunk.
+      last_processed_req_result_ = ResponseResult::kInProgress;
+      RTC_DLOG(LS_VERBOSE) << log_prefix_
+                           << "Reset outgoing; Sender last_assigned="
+                           << *req->sender_last_assigned_tsn()
+                           << " - not yet reached -> InProgress";
+    } else {
+      // https://datatracker.ietf.org/doc/html/rfc6525#section-5.2.2
+      // E3) If no stream numbers are listed in the parameter, then all incoming
+      // streams MUST be reset to 0 as the next expected SSN. If specific stream
+      // numbers are listed, then only these specific streams MUST be reset to
+      // 0, and all other non-listed SSNs remain unchanged. E4: Any queued TSNs
+      // (queued at step E2) MUST now be released and processed normally.
+      reassembly_queue_->ResetStreamsAndLeaveDeferredReset(req->stream_ids());
+      ctx_->callbacks().OnIncomingStreamsReset(req->stream_ids());
+      last_processed_req_result_ = ResponseResult::kSuccessPerformed;
+
+      RTC_DLOG(LS_VERBOSE) << log_prefix_
+                           << "Reset outgoing; Sender last_assigned="
+                           << *req->sender_last_assigned_tsn()
+                           << " - reached -> SuccessPerformed";
+    }
+    responses.push_back(ReconfigurationResponseParameter(
+        req->request_sequence_number(), last_processed_req_result_));
+  }
+}
+
+void StreamResetHandler::HandleResetIncoming(
+    const ParameterDescriptor& descriptor,
+    std::vector<ReconfigurationResponseParameter>& responses) {
+  absl::optional<IncomingSSNResetRequestParameter> req =
+      IncomingSSNResetRequestParameter::Parse(descriptor.data);
+  if (!req.has_value()) {
+    ctx_->callbacks().OnError(ErrorKind::kParseFailed,
+                              "Failed to parse Incoming Reset command");
+    return;
+  }
+
+  UnwrappedReconfigRequestSn request_sn =
+      incoming_reconfig_request_sn_unwrapper_.Unwrap(
+          req->request_sequence_number());
+
+  if (ValidateReqSeqNbr(request_sn, responses)) {
+    responses.push_back(ReconfigurationResponseParameter(
+        req->request_sequence_number(), ResponseResult::kSuccessNothingToDo));
+    last_processed_req_seq_nbr_ = request_sn;
+  }
+}
+
+void StreamResetHandler::HandleResponse(const ParameterDescriptor& descriptor) {
+  absl::optional<ReconfigurationResponseParameter> resp =
+      ReconfigurationResponseParameter::Parse(descriptor.data);
+  if (!resp.has_value()) {
+    ctx_->callbacks().OnError(
+        ErrorKind::kParseFailed,
+        "Failed to parse Reconfiguration Response command");
+    return;
+  }
+
+  if (current_request_.has_value() && current_request_->has_been_sent() &&
+      resp->response_sequence_number() == current_request_->req_seq_nbr()) {
+    reconfig_timer_->Stop();
+
+    switch (resp->result()) {
+      case ResponseResult::kSuccessNothingToDo:
+      case ResponseResult::kSuccessPerformed:
+        RTC_DLOG(LS_VERBOSE)
+            << log_prefix_ << "Reset stream success, req_seq_nbr="
+            << *current_request_->req_seq_nbr() << ", streams="
+            << StrJoin(current_request_->streams(), ",",
+                       [](rtc::StringBuilder& sb, StreamID stream_id) {
+                         sb << *stream_id;
+                       });
+        ctx_->callbacks().OnStreamsResetPerformed(current_request_->streams());
+        current_request_ = absl::nullopt;
+        retransmission_queue_->CommitResetStreams();
+        break;
+      case ResponseResult::kInProgress:
+        RTC_DLOG(LS_VERBOSE)
+            << log_prefix_ << "Reset stream still pending, req_seq_nbr="
+            << *current_request_->req_seq_nbr() << ", streams="
+            << StrJoin(current_request_->streams(), ",",
+                       [](rtc::StringBuilder& sb, StreamID stream_id) {
+                         sb << *stream_id;
+                       });
+        // Force this request to be sent again, but with new req_seq_nbr.
+        current_request_->PrepareRetransmission();
+        reconfig_timer_->set_duration(ctx_->current_rto());
+        reconfig_timer_->Start();
+        break;
+      case ResponseResult::kErrorRequestAlreadyInProgress:
+      case ResponseResult::kDenied:
+      case ResponseResult::kErrorWrongSSN:
+      case ResponseResult::kErrorBadSequenceNumber:
+        RTC_DLOG(LS_WARNING)
+            << log_prefix_ << "Reset stream error=" << ToString(resp->result())
+            << ", req_seq_nbr=" << *current_request_->req_seq_nbr()
+            << ", streams="
+            << StrJoin(current_request_->streams(), ",",
+                       [](rtc::StringBuilder& sb, StreamID stream_id) {
+                         sb << *stream_id;
+                       });
+        ctx_->callbacks().OnStreamsResetFailed(current_request_->streams(),
+                                               ToString(resp->result()));
+        current_request_ = absl::nullopt;
+        retransmission_queue_->RollbackResetStreams();
+        break;
+    }
+  }
+}
+
+absl::optional<ReConfigChunk> StreamResetHandler::MakeStreamResetRequest() {
+  // Only send stream resets if there are streams to reset, and no current
+  // ongoing request (there can only be one at a time), and if the stream
+  // can be reset.
+  if (current_request_.has_value() ||
+      !retransmission_queue_->HasStreamsReadyToBeReset()) {
+    return absl::nullopt;
+  }
+
+  current_request_.emplace(retransmission_queue_->last_assigned_tsn(),
+                           retransmission_queue_->BeginResetStreams());
+  reconfig_timer_->set_duration(ctx_->current_rto());
+  reconfig_timer_->Start();
+  return MakeReconfigChunk();
+}
+
+ReConfigChunk StreamResetHandler::MakeReconfigChunk() {
+  // The req_seq_nbr will be empty if the request has never been sent before,
+  // or if it was sent, but the sender responded "in progress", and then the
+  // req_seq_nbr will be cleared to re-send with a new number. But if the
+  // request is re-sent due to timeout (reconfig-timer expiring), the same
+  // req_seq_nbr will be used.
+  RTC_DCHECK(current_request_.has_value());
+
+  if (!current_request_->has_been_sent()) {
+    current_request_->PrepareToSend(next_outgoing_req_seq_nbr_);
+    next_outgoing_req_seq_nbr_ =
+        ReconfigRequestSN(*next_outgoing_req_seq_nbr_ + 1);
+  }
+
+  Parameters::Builder params_builder =
+      Parameters::Builder().Add(OutgoingSSNResetRequestParameter(
+          current_request_->req_seq_nbr(), current_request_->req_seq_nbr(),
+          current_request_->sender_last_assigned_tsn(),
+          current_request_->streams()));
+
+  return ReConfigChunk(params_builder.Build());
+}
+
+void StreamResetHandler::ResetStreams(
+    rtc::ArrayView<const StreamID> outgoing_streams) {
+  for (StreamID stream_id : outgoing_streams) {
+    retransmission_queue_->PrepareResetStream(stream_id);
+  }
+}
+
+TimeDelta StreamResetHandler::OnReconfigTimerExpiry() {
+  if (current_request_->has_been_sent()) {
+    // There is an outstanding request, which timed out while waiting for a
+    // response.
+    if (!ctx_->IncrementTxErrorCounter("RECONFIG timeout")) {
+      // Timed out. The connection will close after processing the timers.
+      return TimeDelta::Zero();
+    }
+  } else {
+    // There is no outstanding request, but there is a prepared one. This means
+    // that the receiver has previously responded "in progress", which resulted
+    // in retrying the request (but with a new req_seq_nbr) after a while.
+  }
+
+  ctx_->Send(ctx_->PacketBuilder().Add(MakeReconfigChunk()));
+  return ctx_->current_rto();
+}
+
+HandoverReadinessStatus StreamResetHandler::GetHandoverReadiness() const {
+  HandoverReadinessStatus status;
+  if (retransmission_queue_->HasStreamsReadyToBeReset()) {
+    status.Add(HandoverUnreadinessReason::kPendingStreamReset);
+  }
+  if (current_request_.has_value()) {
+    status.Add(HandoverUnreadinessReason::kPendingStreamResetRequest);
+  }
+  return status;
+}
+
+void StreamResetHandler::AddHandoverState(DcSctpSocketHandoverState& state) {
+  state.rx.last_completed_reset_req_sn =
+      last_processed_req_seq_nbr_.Wrap().value();
+  state.tx.next_reset_req_sn = next_outgoing_req_seq_nbr_.value();
+}
+
+}  // namespace dcsctp

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/stream_reset_handler.h

@@ -0,0 +1,238 @@
+/*
+ *  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 NET_DCSCTP_SOCKET_STREAM_RESET_HANDLER_H_
+#define NET_DCSCTP_SOCKET_STREAM_RESET_HANDLER_H_
+
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/bind_front.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/units/time_delta.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/chunk/reconfig_chunk.h"
+#include "net/dcsctp/packet/parameter/incoming_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/outgoing_ssn_reset_request_parameter.h"
+#include "net/dcsctp/packet/parameter/reconfiguration_response_parameter.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/rx/data_tracker.h"
+#include "net/dcsctp/rx/reassembly_queue.h"
+#include "net/dcsctp/socket/context.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/retransmission_queue.h"
+#include "rtc_base/containers/flat_set.h"
+
+namespace dcsctp {
+
+// StreamResetHandler handles sending outgoing stream reset requests (to close
+// an SCTP stream, which translates to closing a data channel).
+//
+// It also handles incoming "outgoing stream reset requests", when the peer
+// wants to close its data channel.
+//
+// Resetting streams is an asynchronous operation where the client will request
+// a request a stream to be reset, but then it might not be performed exactly at
+// this point. First, the sender might need to discard all messages that have
+// been enqueued for this stream, or it may select to wait until all have been
+// sent. At least, it must wait for the currently sending fragmented message to
+// be fully sent, because a stream can't be reset while having received half a
+// message. In the stream reset request, the "sender's last assigned TSN" is
+// provided, which is simply the TSN for which the receiver should've received
+// all messages before this value, before the stream can be reset. Since
+// fragments can get lost or sent out-of-order, the receiver of a request may
+// not have received all the data just yet, and then it will respond to the
+// sender: "In progress". In other words, try again. The sender will then need
+// to start a timer and try the very same request again (but with a new sequence
+// number) until the receiver successfully performs the operation.
+//
+// All this can take some time, and may be driven by timers, so the client will
+// ultimately be notified using callbacks.
+//
+// In this implementation, when a stream is reset, the queued but not-yet-sent
+// messages will be discarded, but that may change in the future. RFC8831 allows
+// both behaviors.
+class StreamResetHandler {
+ public:
+  StreamResetHandler(absl::string_view log_prefix,
+                     Context* context,
+                     TimerManager* timer_manager,
+                     DataTracker* data_tracker,
+                     ReassemblyQueue* reassembly_queue,
+                     RetransmissionQueue* retransmission_queue,
+                     const DcSctpSocketHandoverState* handover_state = nullptr)
+      : log_prefix_(log_prefix),
+        ctx_(context),
+        data_tracker_(data_tracker),
+        reassembly_queue_(reassembly_queue),
+        retransmission_queue_(retransmission_queue),
+        reconfig_timer_(timer_manager->CreateTimer(
+            "re-config",
+            absl::bind_front(&StreamResetHandler::OnReconfigTimerExpiry, this),
+            TimerOptions(webrtc::TimeDelta::Zero()))),
+        next_outgoing_req_seq_nbr_(
+            handover_state
+                ? ReconfigRequestSN(handover_state->tx.next_reset_req_sn)
+                : ReconfigRequestSN(*ctx_->my_initial_tsn())),
+        last_processed_req_seq_nbr_(
+            incoming_reconfig_request_sn_unwrapper_.Unwrap(
+                handover_state
+                    ? ReconfigRequestSN(
+                          handover_state->rx.last_completed_reset_req_sn)
+                    : ReconfigRequestSN(*ctx_->peer_initial_tsn() - 1))),
+        last_processed_req_result_(
+            ReconfigurationResponseParameter::Result::kSuccessNothingToDo) {}
+
+  // Initiates reset of the provided streams. While there can only be one
+  // ongoing stream reset request at any time, this method can be called at any
+  // time and also multiple times. It will enqueue requests that can't be
+  // directly fulfilled, and will asynchronously process them when any ongoing
+  // request has completed.
+  void ResetStreams(rtc::ArrayView<const StreamID> outgoing_streams);
+
+  // Creates a Reset Streams request that must be sent if returned. Will start
+  // the reconfig timer. Will return absl::nullopt if there is no need to
+  // create a request (no streams to reset) or if there already is an ongoing
+  // stream reset request that hasn't completed yet.
+  absl::optional<ReConfigChunk> MakeStreamResetRequest();
+
+  // Called when handling and incoming RE-CONFIG chunk.
+  void HandleReConfig(ReConfigChunk chunk);
+
+  HandoverReadinessStatus GetHandoverReadiness() const;
+
+  void AddHandoverState(DcSctpSocketHandoverState& state);
+
+ private:
+  using UnwrappedReconfigRequestSn = UnwrappedSequenceNumber<ReconfigRequestSN>;
+  // Represents a stream request operation. There can only be one ongoing at
+  // any time, and a sent request may either succeed, fail or result in the
+  // receiver signaling that it can't process it right now, and then it will be
+  // retried.
+  class CurrentRequest {
+   public:
+    CurrentRequest(TSN sender_last_assigned_tsn, std::vector<StreamID> streams)
+        : req_seq_nbr_(absl::nullopt),
+          sender_last_assigned_tsn_(sender_last_assigned_tsn),
+          streams_(std::move(streams)) {}
+
+    // Returns the current request sequence number, if this request has been
+    // sent (check `has_been_sent` first). Will return 0 if the request is just
+    // prepared (or scheduled for retransmission) but not yet sent.
+    ReconfigRequestSN req_seq_nbr() const {
+      return req_seq_nbr_.value_or(ReconfigRequestSN(0));
+    }
+
+    // The sender's last assigned TSN, from the retransmission queue. The
+    // receiver uses this to know when all data up to this TSN has been
+    // received, to know when to safely reset the stream.
+    TSN sender_last_assigned_tsn() const { return sender_last_assigned_tsn_; }
+
+    // The streams that are to be reset.
+    const std::vector<StreamID>& streams() const { return streams_; }
+
+    // If this request has been sent yet. If not, then it's either because it
+    // has only been prepared and not yet sent, or because the received couldn't
+    // apply the request, and then the exact same request will be retried, but
+    // with a new sequence number.
+    bool has_been_sent() const { return req_seq_nbr_.has_value(); }
+
+    // If the receiver can't apply the request yet (and answered "In Progress"),
+    // this will be called to prepare the request to be retransmitted at a later
+    // time.
+    void PrepareRetransmission() { req_seq_nbr_ = absl::nullopt; }
+
+    // If the request hasn't been sent yet, this assigns it a request number.
+    void PrepareToSend(ReconfigRequestSN new_req_seq_nbr) {
+      req_seq_nbr_ = new_req_seq_nbr;
+    }
+
+   private:
+    // If this is set, this request has been sent. If it's not set, the request
+    // has been prepared, but has not yet been sent. This is typically used when
+    // the peer responded "in progress" and the same request (but a different
+    // request number) must be sent again.
+    absl::optional<ReconfigRequestSN> req_seq_nbr_;
+    // The sender's (that's us) last assigned TSN, from the retransmission
+    // queue.
+    TSN sender_last_assigned_tsn_;
+    // The streams that are to be reset in this request.
+    const std::vector<StreamID> streams_;
+  };
+
+  // Called to validate an incoming RE-CONFIG chunk.
+  bool Validate(const ReConfigChunk& chunk);
+
+  // Processes a stream stream reconfiguration chunk and may either return
+  // absl::nullopt (on protocol errors), or a list of responses - either 0, 1
+  // or 2.
+  absl::optional<std::vector<ReconfigurationResponseParameter>> Process(
+      const ReConfigChunk& chunk);
+
+  // Creates the actual RE-CONFIG chunk. A request (which set `current_request`)
+  // must have been created prior.
+  ReConfigChunk MakeReconfigChunk();
+
+  // Called to validate the `req_seq_nbr`, that it's the next in sequence. If it
+  // fails to validate, and returns false, it will also add a response to
+  // `responses`.
+  bool ValidateReqSeqNbr(
+      UnwrappedReconfigRequestSn req_seq_nbr,
+      std::vector<ReconfigurationResponseParameter>& responses);
+
+  // Called when this socket receives an outgoing stream reset request. It might
+  // either be performed straight away, or have to be deferred, and the result
+  // of that will be put in `responses`.
+  void HandleResetOutgoing(
+      const ParameterDescriptor& descriptor,
+      std::vector<ReconfigurationResponseParameter>& responses);
+
+  // Called when this socket receives an incoming stream reset request. This
+  // isn't really supported, but a successful response is put in `responses`.
+  void HandleResetIncoming(
+      const ParameterDescriptor& descriptor,
+      std::vector<ReconfigurationResponseParameter>& responses);
+
+  // Called when receiving a response to an outgoing stream reset request. It
+  // will either commit the stream resetting, if the operation was successful,
+  // or will schedule a retry if it was deferred. And if it failed, the
+  // operation will be rolled back.
+  void HandleResponse(const ParameterDescriptor& descriptor);
+
+  // Expiration handler for the Reconfig timer.
+  webrtc::TimeDelta OnReconfigTimerExpiry();
+
+  const absl::string_view log_prefix_;
+  Context* ctx_;
+  DataTracker* data_tracker_;
+  ReassemblyQueue* reassembly_queue_;
+  RetransmissionQueue* retransmission_queue_;
+  UnwrappedReconfigRequestSn::Unwrapper incoming_reconfig_request_sn_unwrapper_;
+  const std::unique_ptr<Timer> reconfig_timer_;
+
+  // The next sequence number for outgoing stream requests.
+  ReconfigRequestSN next_outgoing_req_seq_nbr_;
+
+  // The current stream request operation.
+  absl::optional<CurrentRequest> current_request_;
+
+  // For incoming requests - last processed request sequence number.
+  UnwrappedReconfigRequestSn last_processed_req_seq_nbr_;
+  // The result from last processed incoming request
+  ReconfigurationResponseParameter::Result last_processed_req_result_;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_STREAM_RESET_HANDLER_H_

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/transmission_control_block.cc

@@ -0,0 +1,338 @@
+/*
+ *  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.
+ */
+#include "net/dcsctp/socket/transmission_control_block.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
+#include "net/dcsctp/packet/chunk/data_chunk.h"
+#include "net/dcsctp/packet/chunk/forward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/idata_chunk.h"
+#include "net/dcsctp/packet/chunk/iforward_tsn_chunk.h"
+#include "net/dcsctp/packet/chunk/reconfig_chunk.h"
+#include "net/dcsctp/packet/chunk/sack_chunk.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/types.h"
+#include "net/dcsctp/rx/data_tracker.h"
+#include "net/dcsctp/rx/reassembly_queue.h"
+#include "net/dcsctp/socket/capabilities.h"
+#include "net/dcsctp/socket/stream_reset_handler.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/retransmission_queue.h"
+#include "net/dcsctp/tx/retransmission_timeout.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace dcsctp {
+using ::webrtc::TimeDelta;
+using ::webrtc::Timestamp;
+
+TransmissionControlBlock::TransmissionControlBlock(
+    TimerManager& timer_manager,
+    absl::string_view log_prefix,
+    const DcSctpOptions& options,
+    const Capabilities& capabilities,
+    DcSctpSocketCallbacks& callbacks,
+    SendQueue& send_queue,
+    VerificationTag my_verification_tag,
+    TSN my_initial_tsn,
+    VerificationTag peer_verification_tag,
+    TSN peer_initial_tsn,
+    size_t a_rwnd,
+    TieTag tie_tag,
+    PacketSender& packet_sender,
+    std::function<bool()> is_connection_established)
+    : log_prefix_(log_prefix),
+      options_(options),
+      timer_manager_(timer_manager),
+      capabilities_(capabilities),
+      callbacks_(callbacks),
+      t3_rtx_(timer_manager_.CreateTimer(
+          "t3-rtx",
+          absl::bind_front(&TransmissionControlBlock::OnRtxTimerExpiry, this),
+          TimerOptions(options.rto_initial.ToTimeDelta(),
+                       TimerBackoffAlgorithm::kExponential,
+                       /*max_restarts=*/absl::nullopt,
+                       options.max_timer_backoff_duration.has_value()
+                           ? options.max_timer_backoff_duration->ToTimeDelta()
+                           : TimeDelta::PlusInfinity()))),
+      delayed_ack_timer_(timer_manager_.CreateTimer(
+          "delayed-ack",
+          absl::bind_front(&TransmissionControlBlock::OnDelayedAckTimerExpiry,
+                           this),
+          TimerOptions(options.delayed_ack_max_timeout.ToTimeDelta(),
+                       TimerBackoffAlgorithm::kExponential,
+                       /*max_restarts=*/0,
+                       /*max_backoff_duration=*/TimeDelta::PlusInfinity(),
+                       webrtc::TaskQueueBase::DelayPrecision::kHigh))),
+      my_verification_tag_(my_verification_tag),
+      my_initial_tsn_(my_initial_tsn),
+      peer_verification_tag_(peer_verification_tag),
+      peer_initial_tsn_(peer_initial_tsn),
+      tie_tag_(tie_tag),
+      is_connection_established_(std::move(is_connection_established)),
+      packet_sender_(packet_sender),
+      rto_(options),
+      tx_error_counter_(log_prefix, options),
+      data_tracker_(log_prefix, delayed_ack_timer_.get(), peer_initial_tsn),
+      reassembly_queue_(log_prefix,
+                        peer_initial_tsn,
+                        options.max_receiver_window_buffer_size,
+                        capabilities.message_interleaving),
+      retransmission_queue_(
+          log_prefix,
+          &callbacks_,
+          my_initial_tsn,
+          a_rwnd,
+          send_queue,
+          absl::bind_front(&TransmissionControlBlock::ObserveRTT, this),
+          [this]() { tx_error_counter_.Clear(); },
+          *t3_rtx_,
+          options,
+          capabilities.partial_reliability,
+          capabilities.message_interleaving),
+      stream_reset_handler_(log_prefix,
+                            this,
+                            &timer_manager,
+                            &data_tracker_,
+                            &reassembly_queue_,
+                            &retransmission_queue_),
+      heartbeat_handler_(log_prefix, options, this, &timer_manager_) {
+  send_queue.EnableMessageInterleaving(capabilities.message_interleaving);
+}
+
+void TransmissionControlBlock::ObserveRTT(TimeDelta rtt) {
+  TimeDelta prev_rto = rto_.rto();
+  rto_.ObserveRTT(rtt);
+  RTC_DLOG(LS_VERBOSE) << log_prefix_ << "new rtt=" << webrtc::ToString(rtt)
+                       << ", srtt=" << webrtc::ToString(rto_.srtt())
+                       << ", rto=" << webrtc::ToString(rto_.rto()) << " ("
+                       << webrtc::ToString(prev_rto) << ")";
+  t3_rtx_->set_duration(rto_.rto());
+
+  TimeDelta delayed_ack_tmo = std::min(
+      rto_.rto() * 0.5, options_.delayed_ack_max_timeout.ToTimeDelta());
+  delayed_ack_timer_->set_duration(delayed_ack_tmo);
+}
+
+TimeDelta TransmissionControlBlock::OnRtxTimerExpiry() {
+  Timestamp now = callbacks_.Now();
+  RTC_DLOG(LS_INFO) << log_prefix_ << "Timer " << t3_rtx_->name()
+                    << " has expired";
+  if (cookie_echo_chunk_.has_value()) {
+    // In the COOKIE_ECHO state, let the T1-COOKIE timer trigger
+    // retransmissions, to avoid having two timers doing that.
+    RTC_DLOG(LS_VERBOSE) << "Not retransmitting as T1-cookie is active.";
+  } else {
+    if (IncrementTxErrorCounter("t3-rtx expired")) {
+      retransmission_queue_.HandleT3RtxTimerExpiry();
+      SendBufferedPackets(now);
+    }
+  }
+  return TimeDelta::Zero();
+}
+
+TimeDelta TransmissionControlBlock::OnDelayedAckTimerExpiry() {
+  data_tracker_.HandleDelayedAckTimerExpiry();
+  MaybeSendSack();
+  return TimeDelta::Zero();
+}
+
+void TransmissionControlBlock::MaybeSendSack() {
+  if (data_tracker_.ShouldSendAck(/*also_if_delayed=*/false)) {
+    SctpPacket::Builder builder = PacketBuilder();
+    builder.Add(
+        data_tracker_.CreateSelectiveAck(reassembly_queue_.remaining_bytes()));
+    Send(builder);
+  }
+}
+
+void TransmissionControlBlock::MaybeSendForwardTsn(SctpPacket::Builder& builder,
+                                                   Timestamp now) {
+  if (now >= limit_forward_tsn_until_ &&
+      retransmission_queue_.ShouldSendForwardTsn(now)) {
+    if (capabilities_.message_interleaving) {
+      builder.Add(retransmission_queue_.CreateIForwardTsn());
+    } else {
+      builder.Add(retransmission_queue_.CreateForwardTsn());
+    }
+    Send(builder);
+    // https://datatracker.ietf.org/doc/html/rfc3758
+    // "IMPLEMENTATION NOTE: An implementation may wish to limit the number of
+    // duplicate FORWARD TSN chunks it sends by ... waiting a full RTT before
+    // sending a duplicate FORWARD TSN."
+    // "Any delay applied to the sending of FORWARD TSN chunk SHOULD NOT exceed
+    // 200ms and MUST NOT exceed 500ms".
+    limit_forward_tsn_until_ =
+        now + std::min(TimeDelta::Millis(200), rto_.srtt());
+  }
+}
+
+void TransmissionControlBlock::MaybeSendFastRetransmit() {
+  if (!retransmission_queue_.has_data_to_be_fast_retransmitted()) {
+    return;
+  }
+
+  // https://datatracker.ietf.org/doc/html/rfc4960#section-7.2.4
+  // "Determine how many of the earliest (i.e., lowest TSN) DATA chunks marked
+  // for retransmission will fit into a single packet, subject to constraint of
+  // the path MTU of the destination transport address to which the packet is
+  // being sent.  Call this value K. Retransmit those K DATA chunks in a single
+  // packet.  When a Fast Retransmit is being performed, the sender SHOULD
+  // ignore the value of cwnd and SHOULD NOT delay retransmission for this
+  // single packet."
+
+  SctpPacket::Builder builder(peer_verification_tag_, options_);
+  auto chunks = retransmission_queue_.GetChunksForFastRetransmit(
+      builder.bytes_remaining());
+  for (auto& [tsn, data] : chunks) {
+    if (capabilities_.message_interleaving) {
+      builder.Add(IDataChunk(tsn, std::move(data), false));
+    } else {
+      builder.Add(DataChunk(tsn, std::move(data), false));
+    }
+  }
+  Send(builder);
+}
+
+void TransmissionControlBlock::SendBufferedPackets(SctpPacket::Builder& builder,
+                                                   Timestamp now) {
+  for (int packet_idx = 0;
+       packet_idx < options_.max_burst && retransmission_queue_.can_send_data();
+       ++packet_idx) {
+    // Only add control chunks to the first packet that is sent, if sending
+    // multiple packets in one go (as allowed by the congestion window).
+    if (packet_idx == 0) {
+      if (cookie_echo_chunk_.has_value()) {
+        // https://tools.ietf.org/html/rfc4960#section-5.1
+        // "The COOKIE ECHO chunk can be bundled with any pending outbound DATA
+        // chunks, but it MUST be the first chunk in the packet..."
+        RTC_DCHECK(builder.empty());
+        builder.Add(*cookie_echo_chunk_);
+      }
+
+      // https://tools.ietf.org/html/rfc4960#section-6
+      // "Before an endpoint transmits a DATA chunk, if any received DATA
+      // chunks have not been acknowledged (e.g., due to delayed ack), the
+      // sender should create a SACK and bundle it with the outbound DATA chunk,
+      // as long as the size of the final SCTP packet does not exceed the
+      // current MTU."
+      if (data_tracker_.ShouldSendAck(/*also_if_delayed=*/true)) {
+        builder.Add(data_tracker_.CreateSelectiveAck(
+            reassembly_queue_.remaining_bytes()));
+      }
+      MaybeSendForwardTsn(builder, now);
+      absl::optional<ReConfigChunk> reconfig =
+          stream_reset_handler_.MakeStreamResetRequest();
+      if (reconfig.has_value()) {
+        builder.Add(*reconfig);
+      }
+    }
+
+    auto chunks =
+        retransmission_queue_.GetChunksToSend(now, builder.bytes_remaining());
+    for (auto& [tsn, data] : chunks) {
+      if (capabilities_.message_interleaving) {
+        builder.Add(IDataChunk(tsn, std::move(data), false));
+      } else {
+        builder.Add(DataChunk(tsn, std::move(data), false));
+      }
+    }
+
+    // https://www.ietf.org/archive/id/draft-tuexen-tsvwg-sctp-zero-checksum-02.html#section-4.2
+    // "When an end point sends a packet containing a COOKIE ECHO chunk, it MUST
+    // include a correct CRC32c checksum in the packet containing the COOKIE
+    // ECHO chunk."
+    bool write_checksum =
+        !capabilities_.zero_checksum || cookie_echo_chunk_.has_value();
+    if (!packet_sender_.Send(builder, write_checksum)) {
+      break;
+    }
+
+    if (cookie_echo_chunk_.has_value()) {
+      // https://tools.ietf.org/html/rfc4960#section-5.1
+      // "...  until the COOKIE ACK is returned the sender MUST NOT send any
+      // other packets to the peer."
+      break;
+    }
+  }
+}
+
+std::string TransmissionControlBlock::ToString() const {
+  rtc::StringBuilder sb;
+
+  sb.AppendFormat(
+      "verification_tag=%08x, last_cumulative_ack=%u, capabilities=",
+      *peer_verification_tag_, *data_tracker_.last_cumulative_acked_tsn());
+
+  if (capabilities_.partial_reliability) {
+    sb << "PR,";
+  }
+  if (capabilities_.message_interleaving) {
+    sb << "IL,";
+  }
+  if (capabilities_.reconfig) {
+    sb << "Reconfig,";
+  }
+  if (capabilities_.zero_checksum) {
+    sb << "ZeroChecksum,";
+  }
+  sb << " max_in=" << capabilities_.negotiated_maximum_incoming_streams;
+  sb << " max_out=" << capabilities_.negotiated_maximum_outgoing_streams;
+
+  return sb.Release();
+}
+
+HandoverReadinessStatus TransmissionControlBlock::GetHandoverReadiness() const {
+  HandoverReadinessStatus status;
+  status.Add(data_tracker_.GetHandoverReadiness());
+  status.Add(stream_reset_handler_.GetHandoverReadiness());
+  status.Add(reassembly_queue_.GetHandoverReadiness());
+  status.Add(retransmission_queue_.GetHandoverReadiness());
+  return status;
+}
+
+void TransmissionControlBlock::AddHandoverState(
+    DcSctpSocketHandoverState& state) {
+  state.capabilities.partial_reliability = capabilities_.partial_reliability;
+  state.capabilities.message_interleaving = capabilities_.message_interleaving;
+  state.capabilities.reconfig = capabilities_.reconfig;
+  state.capabilities.zero_checksum = capabilities_.zero_checksum;
+  state.capabilities.negotiated_maximum_incoming_streams =
+      capabilities_.negotiated_maximum_incoming_streams;
+  state.capabilities.negotiated_maximum_outgoing_streams =
+      capabilities_.negotiated_maximum_outgoing_streams;
+
+  state.my_verification_tag = my_verification_tag().value();
+  state.peer_verification_tag = peer_verification_tag().value();
+  state.my_initial_tsn = my_initial_tsn().value();
+  state.peer_initial_tsn = peer_initial_tsn().value();
+  state.tie_tag = tie_tag().value();
+
+  data_tracker_.AddHandoverState(state);
+  stream_reset_handler_.AddHandoverState(state);
+  reassembly_queue_.AddHandoverState(state);
+  retransmission_queue_.AddHandoverState(state);
+}
+
+void TransmissionControlBlock::RestoreFromState(
+    const DcSctpSocketHandoverState& state) {
+  data_tracker_.RestoreFromState(state);
+  retransmission_queue_.RestoreFromState(state);
+  reassembly_queue_.RestoreFromState(state);
+}
+}  // namespace dcsctp

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/transmission_control_block.h

@@ -0,0 +1,194 @@
+/*
+ *  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 NET_DCSCTP_SOCKET_TRANSMISSION_CONTROL_BLOCK_H_
+#define NET_DCSCTP_SOCKET_TRANSMISSION_CONTROL_BLOCK_H_
+
+#include <cstdint>
+#include <functional>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/bind_front.h"
+#include "absl/strings/string_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "net/dcsctp/common/sequence_numbers.h"
+#include "net/dcsctp/packet/chunk/cookie_echo_chunk.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/rx/data_tracker.h"
+#include "net/dcsctp/rx/reassembly_queue.h"
+#include "net/dcsctp/socket/capabilities.h"
+#include "net/dcsctp/socket/context.h"
+#include "net/dcsctp/socket/heartbeat_handler.h"
+#include "net/dcsctp/socket/packet_sender.h"
+#include "net/dcsctp/socket/stream_reset_handler.h"
+#include "net/dcsctp/timer/timer.h"
+#include "net/dcsctp/tx/retransmission_error_counter.h"
+#include "net/dcsctp/tx/retransmission_queue.h"
+#include "net/dcsctp/tx/retransmission_timeout.h"
+#include "net/dcsctp/tx/send_queue.h"
+
+namespace dcsctp {
+
+// The TransmissionControlBlock (TCB) represents an open connection to a peer,
+// and holds all the resources for that. If the connection is e.g. shutdown,
+// closed or restarted, this object will be deleted and/or replaced.
+class TransmissionControlBlock : public Context {
+ public:
+  TransmissionControlBlock(TimerManager& timer_manager,
+                           absl::string_view log_prefix,
+                           const DcSctpOptions& options,
+                           const Capabilities& capabilities,
+                           DcSctpSocketCallbacks& callbacks,
+                           SendQueue& send_queue,
+                           VerificationTag my_verification_tag,
+                           TSN my_initial_tsn,
+                           VerificationTag peer_verification_tag,
+                           TSN peer_initial_tsn,
+                           size_t a_rwnd,
+                           TieTag tie_tag,
+                           PacketSender& packet_sender,
+                           std::function<bool()> is_connection_established);
+
+  // Implementation of `Context`.
+  bool is_connection_established() const override {
+    return is_connection_established_();
+  }
+  TSN my_initial_tsn() const override { return my_initial_tsn_; }
+  TSN peer_initial_tsn() const override { return peer_initial_tsn_; }
+  DcSctpSocketCallbacks& callbacks() const override { return callbacks_; }
+  void ObserveRTT(webrtc::TimeDelta rtt) override;
+  webrtc::TimeDelta current_rto() const override { return rto_.rto(); }
+  bool IncrementTxErrorCounter(absl::string_view reason) override {
+    return tx_error_counter_.Increment(reason);
+  }
+  void ClearTxErrorCounter() override { tx_error_counter_.Clear(); }
+  SctpPacket::Builder PacketBuilder() const override {
+    return SctpPacket::Builder(peer_verification_tag_, options_);
+  }
+  bool HasTooManyTxErrors() const override {
+    return tx_error_counter_.IsExhausted();
+  }
+  void Send(SctpPacket::Builder& builder) override {
+    packet_sender_.Send(builder,
+                        /*write_checksum=*/!capabilities_.zero_checksum);
+  }
+
+  // Other accessors
+  DataTracker& data_tracker() { return data_tracker_; }
+  ReassemblyQueue& reassembly_queue() { return reassembly_queue_; }
+  RetransmissionQueue& retransmission_queue() { return retransmission_queue_; }
+  StreamResetHandler& stream_reset_handler() { return stream_reset_handler_; }
+  HeartbeatHandler& heartbeat_handler() { return heartbeat_handler_; }
+  size_t cwnd() const { return retransmission_queue_.cwnd(); }
+  webrtc::TimeDelta current_srtt() const { return rto_.srtt(); }
+
+  // Returns this socket's verification tag, set in all packet headers.
+  VerificationTag my_verification_tag() const { return my_verification_tag_; }
+  // Returns the peer's verification tag, which should be in received packets.
+  VerificationTag peer_verification_tag() const {
+    return peer_verification_tag_;
+  }
+  // All negotiated supported capabilities.
+  const Capabilities& capabilities() const { return capabilities_; }
+  // A 64-bit tie-tag, used to e.g. detect reconnections.
+  TieTag tie_tag() const { return tie_tag_; }
+
+  // Sends a SACK, if there is a need to.
+  void MaybeSendSack();
+
+  // Sends a FORWARD-TSN, if it is needed and allowed (rate-limited).
+  void MaybeSendForwardTsn(SctpPacket::Builder& builder, webrtc::Timestamp now);
+
+  // Will be set while the socket is in kCookieEcho state. In this state, there
+  // can only be a single packet outstanding, and it must contain the COOKIE
+  // ECHO chunk as the first chunk in that packet, until the COOKIE ACK has been
+  // received, which will make the socket call `ClearCookieEchoChunk`.
+  void SetCookieEchoChunk(CookieEchoChunk chunk) {
+    cookie_echo_chunk_ = std::move(chunk);
+  }
+
+  // Called when the COOKIE ACK chunk has been received, to allow further
+  // packets to be sent.
+  void ClearCookieEchoChunk() { cookie_echo_chunk_ = absl::nullopt; }
+
+  bool has_cookie_echo_chunk() const { return cookie_echo_chunk_.has_value(); }
+
+  void MaybeSendFastRetransmit();
+
+  // Fills `builder` (which may already be filled with control chunks) with
+  // other control and data chunks, and sends packets as much as can be
+  // allowed by the congestion control algorithm.
+  void SendBufferedPackets(SctpPacket::Builder& builder, webrtc::Timestamp now);
+
+  // As above, but without passing in a builder. If `cookie_echo_chunk_` is
+  // present, then only one packet will be sent, with this chunk as the first
+  // chunk.
+  void SendBufferedPackets(webrtc::Timestamp now) {
+    SctpPacket::Builder builder(peer_verification_tag_, options_);
+    SendBufferedPackets(builder, now);
+  }
+
+  // Returns a textual representation of this object, for logging.
+  std::string ToString() const;
+
+  HandoverReadinessStatus GetHandoverReadiness() const;
+
+  void AddHandoverState(DcSctpSocketHandoverState& state);
+  void RestoreFromState(const DcSctpSocketHandoverState& handover_state);
+
+ private:
+  // Will be called when the retransmission timer (t3-rtx) expires.
+  webrtc::TimeDelta OnRtxTimerExpiry();
+  // Will be called when the delayed ack timer expires.
+  webrtc::TimeDelta OnDelayedAckTimerExpiry();
+
+  const absl::string_view log_prefix_;
+  const DcSctpOptions options_;
+  TimerManager& timer_manager_;
+  // Negotiated capabilities that both peers support.
+  const Capabilities capabilities_;
+  DcSctpSocketCallbacks& callbacks_;
+  // The data retransmission timer, called t3-rtx in SCTP.
+  const std::unique_ptr<Timer> t3_rtx_;
+  // Delayed ack timer, which triggers when acks should be sent (when delayed).
+  const std::unique_ptr<Timer> delayed_ack_timer_;
+  const VerificationTag my_verification_tag_;
+  const TSN my_initial_tsn_;
+  const VerificationTag peer_verification_tag_;
+  const TSN peer_initial_tsn_;
+  // Nonce, used to detect reconnections.
+  const TieTag tie_tag_;
+  const std::function<bool()> is_connection_established_;
+  PacketSender& packet_sender_;
+  // Rate limiting of FORWARD-TSN. Next can be sent at or after this timestamp.
+  webrtc::Timestamp limit_forward_tsn_until_ = webrtc::Timestamp::Zero();
+
+  RetransmissionTimeout rto_;
+  RetransmissionErrorCounter tx_error_counter_;
+  DataTracker data_tracker_;
+  ReassemblyQueue reassembly_queue_;
+  RetransmissionQueue retransmission_queue_;
+  StreamResetHandler stream_reset_handler_;
+  HeartbeatHandler heartbeat_handler_;
+
+  // Only valid when the socket state == State::kCookieEchoed. In this state,
+  // the socket must wait for COOKIE ACK to continue sending any packets (not
+  // including a COOKIE ECHO). So if `cookie_echo_chunk_` is present, the
+  // SendBufferedChunks will always only just send one packet, with this chunk
+  // as the first chunk in the packet.
+  absl::optional<CookieEchoChunk> cookie_echo_chunk_ = absl::nullopt;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_TRANSMISSION_CONTROL_BLOCK_H_