/* * 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 #include #include #include #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 MaxRetransmits max_retransmissions = MaxRetransmits::NoLimit(); TimeMs expires_at = TimeMs::InfiniteFuture(); // 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 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 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 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 #endif // NET_DCSCTP_TX_SEND_QUEUE_H_