blob: d0d834c901addcb5f44927156c43e6e2c369f2d0 [file] [log] [blame]
* 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 <cstdint>
#include <limits>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/units/timestamp.h"
#include "net/dcsctp/common/internal_types.h"
#include "net/dcsctp/packet/data.h"
#include "net/dcsctp/public/types.h"
namespace dcsctp {
class SendQueue {
// Container for a data chunk that is produced by the SendQueue
struct DataToSend {
DataToSend(OutgoingMessageId message_id, Data data)
: message_id(message_id), data(std::move(data)) {}
OutgoingMessageId message_id;
// The data to send, including all parameters.
Data data;
// Partial reliability - RFC3758
MaxRetransmits max_retransmissions = MaxRetransmits::NoLimit();
webrtc::Timestamp expires_at = webrtc::Timestamp::PlusInfinity();
// Lifecycle - set for the last fragment, and `LifecycleId::NotSet()` for
// all other fragments.
LifecycleId lifecycle_id = LifecycleId::NotSet();
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(webrtc::Timestamp now,
size_t max_size) = 0;
// Discards a partially sent message identified by the parameters
// `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(StreamID stream_id, OutgoingMessageId message_id) = 0;
// Prepares the stream 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
// stream and pauses that stream 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 PrepareResetStream(StreamID stream_id) = 0;
// Indicates if there are any streams that are ready to be reset.
virtual bool HasStreamsReadyToBeReset() const = 0;
// Returns a list of streams that are ready to be included in an outgoing
// stream reset request. Any streams that are returned here must be included
// in an outgoing stream reset request, and there must not be concurrent
// requests. Before calling this method again, you must have called
virtual std::vector<StreamID> GetStreamsReadyToBeReset() = 0;
// Called to commit to reset the streams returned by
// `GetStreamsReadyToBeReset`. 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 returned by
// `GetStreamsReadyToBeReset`. 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;
// Configures the send queue to support interleaved message sending as
// described in RFC8260. Every send queue starts with this value set as
// disabled, but can later change it when the capabilities of the connection
// have been negotiated. This affects the behavior of the `Produce` method.
virtual void EnableMessageInterleaving(bool enabled) = 0;
} // namespace dcsctp