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diff --git a/net/dcsctp/tx/send_queue.h b/net/dcsctp/tx/send_queue.h
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+/*
+ *  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_TX_SEND_QUEUE_H_
+#define NET_DCSCTP_TX_SEND_QUEUE_H_
+
+#include <cstdint>
+#include <limits>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "net/dcsctp/common/internal_types.h"
+#include "net/dcsctp/packet/data.h"
+#include "net/dcsctp/public/types.h"
+
+namespace dcsctp {
+
+class SendQueue {
+ public:
+  // Container for a data chunk that is produced by the SendQueue
+  struct DataToSend {
+    explicit DataToSend(Data data) : data(std::move(data)) {}
+    // The data to send, including all parameters.
+    Data data;
+
+    // Partial reliability - RFC3758
+    absl::optional<int> max_retransmissions;
+    absl::optional<TimeMs> expires_at;
+  };
+
+  virtual ~SendQueue() = default;
+
+  // TODO(boivie): This interface is obviously missing an "Add" function, but
+  // that is postponed a bit until the story around how to model message
+  // prioritization, which is important for any advanced stream scheduler, is
+  // further clarified.
+
+  // Produce a chunk to be sent.
+  //
+  // `max_size` refers to how many payload bytes that may be produced, not
+  // including any headers.
+  virtual absl::optional<DataToSend> Produce(TimeMs now, size_t max_size) = 0;
+
+  // Discards a partially sent message identified by the parameters `unordered`,
+  // `stream_id` and `message_id`. The `message_id` comes from the returned
+  // information when having called `Produce`. A partially sent message means
+  // that it has had at least one fragment of it returned when `Produce` was
+  // called prior to calling this method).
+  //
+  // This is used when a message has been found to be expired (by the partial
+  // reliability extension), and the retransmission queue will signal the
+  // receiver that any partially received message fragments should be skipped.
+  // This means that any remaining fragments in the Send Queue must be removed
+  // as well so that they are not sent.
+  //
+  // This function returns true if this message had unsent fragments still in
+  // the queue that were discarded, and false if there were no such fragments.
+  virtual bool Discard(IsUnordered unordered,
+                       StreamID stream_id,
+                       MID message_id) = 0;
+
+  // Prepares the streams to be reset. This is used to close a WebRTC data
+  // channel and will be signaled to the other side.
+  //
+  // Concretely, it discards all whole (not partly sent) messages in the given
+  // streams and pauses those streams so that future added messages aren't
+  // produced until `ResumeStreams` is called.
+  //
+  // TODO(boivie): Investigate if it really should discard any message at all.
+  // RFC8831 only mentions that "[RFC6525] also guarantees that all the messages
+  // are delivered (or abandoned) before the stream is reset."
+  //
+  // This method can be called multiple times to add more streams to be
+  // reset, and paused while they are resetting. This is the first part of the
+  // two-phase commit protocol to reset streams, where the caller completes the
+  // procedure by either calling `CommitResetStreams` or `RollbackResetStreams`.
+  virtual void PrepareResetStreams(rtc::ArrayView<const StreamID> streams) = 0;
+
+  // Returns true if all non-discarded messages during `PrepareResetStreams`
+  // (which are those that was partially sent before that method was called)
+  // have been sent.
+  virtual bool CanResetStreams() const = 0;
+
+  // Called to commit to reset the streams provided to `PrepareResetStreams`.
+  // It will reset the stream sequence numbers (SSNs) and message identifiers
+  // (MIDs) and resume the paused streams.
+  virtual void CommitResetStreams() = 0;
+
+  // Called to abort the resetting of streams provided to `PrepareResetStreams`.
+  // Will resume the paused streams without resetting the stream sequence
+  // numbers (SSNs) or message identifiers (MIDs). Note that the non-partial
+  // messages that were discarded when calling `PrepareResetStreams` will not be
+  // recovered, to better match the intention from the sender to "close the
+  // channel".
+  virtual void RollbackResetStreams() = 0;
+
+  // Resets all message identifier counters (MID, SSN) and makes all partially
+  // messages be ready to be re-sent in full. This is used when the peer has
+  // been detected to have restarted and is used to try to minimize the amount
+  // of data loss. However, data loss cannot be completely guaranteed when a
+  // peer restarts.
+  virtual void Reset() = 0;
+
+  // Returns the amount of buffered data. This doesn't include packets that are
+  // e.g. inflight.
+  virtual size_t buffered_amount(StreamID stream_id) const = 0;
+
+  // Returns the total amount of buffer data, for all streams.
+  virtual size_t total_buffered_amount() const = 0;
+
+  // Returns the limit for the `OnBufferedAmountLow` event. Default value is 0.
+  virtual size_t buffered_amount_low_threshold(StreamID stream_id) const = 0;
+
+  // Sets a limit for the `OnBufferedAmountLow` event.
+  virtual void SetBufferedAmountLowThreshold(StreamID stream_id,
+                                             size_t bytes) = 0;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_TX_SEND_QUEUE_H_