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Fr4nz D13trich 2025-11-22 14:04:28 +01:00
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68
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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.
*/
#include "net/dcsctp/public/dcsctp_handover_state.h"
#include <string>
#include "absl/strings/string_view.h"
namespace dcsctp {
namespace {
constexpr absl::string_view HandoverUnreadinessReasonToString(
HandoverUnreadinessReason reason) {
switch (reason) {
case HandoverUnreadinessReason::kWrongConnectionState:
return "WRONG_CONNECTION_STATE";
case HandoverUnreadinessReason::kSendQueueNotEmpty:
return "SEND_QUEUE_NOT_EMPTY";
case HandoverUnreadinessReason::kDataTrackerTsnBlocksPending:
return "DATA_TRACKER_TSN_BLOCKS_PENDING";
case HandoverUnreadinessReason::kReassemblyQueueDeliveredTSNsGap:
return "REASSEMBLY_QUEUE_DELIVERED_TSN_GAP";
case HandoverUnreadinessReason::kStreamResetDeferred:
return "STREAM_RESET_DEFERRED";
case HandoverUnreadinessReason::kOrderedStreamHasUnassembledChunks:
return "ORDERED_STREAM_HAS_UNASSEMBLED_CHUNKS";
case HandoverUnreadinessReason::kUnorderedStreamHasUnassembledChunks:
return "UNORDERED_STREAM_HAS_UNASSEMBLED_CHUNKS";
case HandoverUnreadinessReason::kRetransmissionQueueOutstandingData:
return "RETRANSMISSION_QUEUE_OUTSTANDING_DATA";
case HandoverUnreadinessReason::kRetransmissionQueueFastRecovery:
return "RETRANSMISSION_QUEUE_FAST_RECOVERY";
case HandoverUnreadinessReason::kRetransmissionQueueNotEmpty:
return "RETRANSMISSION_QUEUE_NOT_EMPTY";
case HandoverUnreadinessReason::kPendingStreamReset:
return "PENDING_STREAM_RESET";
case HandoverUnreadinessReason::kPendingStreamResetRequest:
return "PENDING_STREAM_RESET_REQUEST";
}
}
} // namespace
std::string HandoverReadinessStatus::ToString() const {
std::string result;
for (uint32_t bit = 1;
bit <= static_cast<uint32_t>(HandoverUnreadinessReason::kMax);
bit *= 2) {
auto flag = static_cast<HandoverUnreadinessReason>(bit);
if (Contains(flag)) {
if (!result.empty()) {
result.append(",");
}
absl::string_view s = HandoverUnreadinessReasonToString(flag);
result.append(s.data(), s.size());
}
}
if (result.empty()) {
result = "READY";
}
return result;
}
} // namespace dcsctp

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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_PUBLIC_DCSCTP_HANDOVER_STATE_H_
#define NET_DCSCTP_PUBLIC_DCSCTP_HANDOVER_STATE_H_
#include <cstdint>
#include <string>
#include <vector>
#include "rtc_base/strong_alias.h"
namespace dcsctp {
// Stores state snapshot of a dcSCTP socket. The snapshot can be used to
// recreate the socket - possibly in another process. This state should be
// treaded as opaque - the calling client should not inspect or alter it except
// for serialization. Serialization is not provided by dcSCTP. If needed it has
// to be implemented in the calling client.
struct DcSctpSocketHandoverState {
enum class SocketState {
kClosed,
kConnected,
};
SocketState socket_state = SocketState::kClosed;
uint32_t my_verification_tag = 0;
uint32_t my_initial_tsn = 0;
uint32_t peer_verification_tag = 0;
uint32_t peer_initial_tsn = 0;
uint64_t tie_tag = 0;
struct Capabilities {
bool partial_reliability = false;
bool message_interleaving = false;
bool reconfig = false;
bool zero_checksum = false;
uint16_t negotiated_maximum_incoming_streams = 0;
uint16_t negotiated_maximum_outgoing_streams = 0;
};
Capabilities capabilities;
struct OutgoingStream {
uint32_t id = 0;
uint32_t next_ssn = 0;
uint32_t next_unordered_mid = 0;
uint32_t next_ordered_mid = 0;
uint16_t priority = 0;
};
struct Transmission {
uint32_t next_tsn = 0;
uint32_t next_reset_req_sn = 0;
uint32_t cwnd = 0;
uint32_t rwnd = 0;
uint32_t ssthresh = 0;
uint32_t partial_bytes_acked = 0;
std::vector<OutgoingStream> streams;
};
Transmission tx;
struct OrderedStream {
uint32_t id = 0;
uint32_t next_ssn = 0;
};
struct UnorderedStream {
uint32_t id = 0;
};
struct Receive {
bool seen_packet = false;
uint32_t last_cumulative_acked_tsn = 0;
uint32_t last_assembled_tsn = 0;
uint32_t last_completed_deferred_reset_req_sn = 0;
uint32_t last_completed_reset_req_sn = 0;
std::vector<OrderedStream> ordered_streams;
std::vector<UnorderedStream> unordered_streams;
};
Receive rx;
};
// A list of possible reasons for a socket to be not ready for handover.
enum class HandoverUnreadinessReason : uint32_t {
kWrongConnectionState = 1,
kSendQueueNotEmpty = 2,
kPendingStreamResetRequest = 4,
kDataTrackerTsnBlocksPending = 8,
kPendingStreamReset = 16,
kReassemblyQueueDeliveredTSNsGap = 32,
kStreamResetDeferred = 64,
kOrderedStreamHasUnassembledChunks = 128,
kUnorderedStreamHasUnassembledChunks = 256,
kRetransmissionQueueOutstandingData = 512,
kRetransmissionQueueFastRecovery = 1024,
kRetransmissionQueueNotEmpty = 2048,
kMax = kRetransmissionQueueNotEmpty,
};
// Return value of `DcSctpSocketInterface::GetHandoverReadiness`. Set of
// `HandoverUnreadinessReason` bits. When no bit is set, the socket is in the
// state in which a snapshot of the state can be made by
// `GetHandoverStateAndClose()`.
class HandoverReadinessStatus
: public webrtc::StrongAlias<class HandoverReadinessStatusTag, uint32_t> {
public:
// Constructs an empty `HandoverReadinessStatus` which represents ready state.
constexpr HandoverReadinessStatus()
: webrtc::StrongAlias<class HandoverReadinessStatusTag, uint32_t>(0) {}
// Constructs status object that contains a single reason for not being
// handover ready.
constexpr explicit HandoverReadinessStatus(HandoverUnreadinessReason reason)
: webrtc::StrongAlias<class HandoverReadinessStatusTag, uint32_t>(
static_cast<uint32_t>(reason)) {}
// Convenience methods
constexpr bool IsReady() const { return value() == 0; }
constexpr bool Contains(HandoverUnreadinessReason reason) const {
return value() & static_cast<uint32_t>(reason);
}
HandoverReadinessStatus& Add(HandoverUnreadinessReason reason) {
return Add(HandoverReadinessStatus(reason));
}
HandoverReadinessStatus& Add(HandoverReadinessStatus status) {
value() |= status.value();
return *this;
}
std::string ToString() const;
};
} // namespace dcsctp
#endif // NET_DCSCTP_PUBLIC_DCSCTP_HANDOVER_STATE_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_PUBLIC_DCSCTP_MESSAGE_H_
#define NET_DCSCTP_PUBLIC_DCSCTP_MESSAGE_H_
#include <cstdint>
#include <utility>
#include <vector>
#include "api/array_view.h"
#include "net/dcsctp/public/types.h"
namespace dcsctp {
// An SCTP message is a group of bytes sent and received as a whole on a
// specified stream identifier (`stream_id`), and with a payload protocol
// identifier (`ppid`).
class DcSctpMessage {
public:
DcSctpMessage(StreamID stream_id, PPID ppid, std::vector<uint8_t> payload)
: stream_id_(stream_id), ppid_(ppid), payload_(std::move(payload)) {}
DcSctpMessage(DcSctpMessage&& other) = default;
DcSctpMessage& operator=(DcSctpMessage&& other) = default;
DcSctpMessage(const DcSctpMessage&) = delete;
DcSctpMessage& operator=(const DcSctpMessage&) = delete;
// The stream identifier to which the message is sent.
StreamID stream_id() const { return stream_id_; }
// The payload protocol identifier (ppid) associated with the message.
PPID ppid() const { return ppid_; }
// The payload of the message.
rtc::ArrayView<const uint8_t> payload() const { return payload_; }
// When destructing the message, extracts the payload.
std::vector<uint8_t> ReleasePayload() && { return std::move(payload_); }
private:
StreamID stream_id_;
PPID ppid_;
std::vector<uint8_t> payload_;
};
} // namespace dcsctp
#endif // NET_DCSCTP_PUBLIC_DCSCTP_MESSAGE_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_PUBLIC_DCSCTP_OPTIONS_H_
#define NET_DCSCTP_PUBLIC_DCSCTP_OPTIONS_H_
#include <stddef.h>
#include <stdint.h>
#include "absl/types/optional.h"
#include "net/dcsctp/public/types.h"
namespace dcsctp {
struct DcSctpOptions {
// The largest safe SCTP packet. Starting from the minimum guaranteed MTU
// value of 1280 for IPv6 (which may not support fragmentation), take off 85
// bytes for DTLS/TURN/TCP/IP and ciphertext overhead.
//
// Additionally, it's possible that TURN adds an additional 4 bytes of
// overhead after a channel has been established, so an additional 4 bytes is
// subtracted
//
// 1280 IPV6 MTU
// -40 IPV6 header
// -8 UDP
// -24 GCM Cipher
// -13 DTLS record header
// -4 TURN ChannelData
// = 1191 bytes.
static constexpr size_t kMaxSafeMTUSize = 1191;
// The local port for which the socket is supposed to be bound to. Incoming
// packets will be verified that they are sent to this port number and all
// outgoing packets will have this port number as source port.
int local_port = 5000;
// The remote port to send packets to. All outgoing packets will have this
// port number as destination port.
int remote_port = 5000;
// The announced maximum number of incoming streams. Note that this value is
// constant and can't be currently increased in run-time as "Add Incoming
// Streams Request" in RFC6525 isn't supported.
//
// The socket implementation doesn't have any per-stream fixed costs, which is
// why the default value is set to be the maximum value.
uint16_t announced_maximum_incoming_streams = 65535;
// The announced maximum number of outgoing streams. Note that this value is
// constant and can't be currently increased in run-time as "Add Outgoing
// Streams Request" in RFC6525 isn't supported.
//
// The socket implementation doesn't have any per-stream fixed costs, which is
// why the default value is set to be the maximum value.
uint16_t announced_maximum_outgoing_streams = 65535;
// Maximum SCTP packet size. The library will limit the size of generated
// packets to be less than or equal to this number. This does not include any
// overhead of DTLS, TURN, UDP or IP headers.
size_t mtu = kMaxSafeMTUSize;
// The largest allowed message payload to be sent. Messages will be rejected
// if their payload is larger than this value. Note that this doesn't affect
// incoming messages, which may larger than this value (but smaller than
// `max_receiver_window_buffer_size`).
size_t max_message_size = 256 * 1024;
// The default stream priority, if not overridden by
// `SctpSocket::SetStreamPriority`. The default value is selected to be
// compatible with https://www.w3.org/TR/webrtc-priority/, section 4.2-4.3.
StreamPriority default_stream_priority = StreamPriority(256);
// Maximum received window buffer size. This should be a bit larger than the
// largest sized message you want to be able to receive. This essentially
// limits the memory usage on the receive side. Note that memory is allocated
// dynamically, and this represents the maximum amount of buffered data. The
// actual memory usage of the library will be smaller in normal operation, and
// will be larger than this due to other allocations and overhead if the
// buffer is fully utilized.
size_t max_receiver_window_buffer_size = 5 * 1024 * 1024;
// Maximum send buffer size. It will not be possible to queue more data than
// this before sending it.
size_t max_send_buffer_size = 2'000'000;
// A threshold that, when the amount of data in the send buffer goes below
// this value, will trigger `DcSctpCallbacks::OnTotalBufferedAmountLow`.
size_t total_buffered_amount_low_threshold = 1'800'000;
// Max allowed RTT value. When the RTT is measured and it's found to be larger
// than this value, it will be discarded and not used for e.g. any RTO
// calculation. The default value is an extreme maximum but can be adapted
// to better match the environment.
DurationMs rtt_max = DurationMs(60'000);
// Initial RTO value.
DurationMs rto_initial = DurationMs(500);
// Maximum RTO value.
DurationMs rto_max = DurationMs(60'000);
// Minimum RTO value. This must be larger than an expected peer delayed ack
// timeout.
DurationMs rto_min = DurationMs(400);
// T1-init timeout.
DurationMs t1_init_timeout = DurationMs(1000);
// T1-cookie timeout.
DurationMs t1_cookie_timeout = DurationMs(1000);
// T2-shutdown timeout.
DurationMs t2_shutdown_timeout = DurationMs(1000);
// For t1-init, t1-cookie, t2-shutdown, t3-rtx, this value - if set - will be
// the upper bound on how large the exponentially backed off timeout can
// become. The lower the duration, the faster the connection can recover on
// transient network issues. Setting this value may require changing
// `max_retransmissions` and `max_init_retransmits` to ensure that the
// connection is not closed too quickly.
absl::optional<DurationMs> max_timer_backoff_duration = absl::nullopt;
// Hearbeat interval (on idle connections only). Set to zero to disable.
DurationMs heartbeat_interval = DurationMs(30000);
// The maximum time when a SACK will be sent from the arrival of an
// unacknowledged packet. Whatever is smallest of RTO/2 and this will be used.
DurationMs delayed_ack_max_timeout = DurationMs(200);
// The minimum limit for the measured RTT variance
//
// Setting this below the expected delayed ack timeout (+ margin) of the peer
// might result in unnecessary retransmissions, as the maximum time it takes
// to ACK a DATA chunk is typically RTT + ATO (delayed ack timeout), and when
// the SCTP channel is quite idle, and heartbeats dominate the source of RTT
// measurement, the RTO would converge with the smoothed RTT (SRTT). The
// default ATO is 200ms in usrsctp, and a 20ms (10%) margin would include the
// processing time of received packets and the clock granularity when setting
// the delayed ack timer on the peer.
//
// This is described for TCP in
// https://datatracker.ietf.org/doc/html/rfc6298#section-4.
DurationMs min_rtt_variance = DurationMs(220);
// The initial congestion window size, in number of MTUs.
// See https://tools.ietf.org/html/rfc4960#section-7.2.1 which defaults at ~3
// and https://research.google/pubs/pub36640/ which argues for at least ten
// segments.
size_t cwnd_mtus_initial = 10;
// The minimum congestion window size, in number of MTUs, upon detection of
// packet loss by SACK. Note that if the retransmission timer expires, the
// congestion window will be as small as one MTU. See
// https://tools.ietf.org/html/rfc4960#section-7.2.3.
size_t cwnd_mtus_min = 4;
// When the congestion window is at or above this number of MTUs, the
// congestion control algorithm will avoid filling the congestion window
// fully, if that results in fragmenting large messages into quite small
// packets. When the congestion window is smaller than this option, it will
// aim to fill the congestion window as much as it can, even if it results in
// creating small fragmented packets.
size_t avoid_fragmentation_cwnd_mtus = 6;
// The number of packets that may be sent at once. This is limited to avoid
// bursts that too quickly fill the send buffer. Typically in a a socket in
// its "slow start" phase (when it sends as much as it can), it will send
// up to three packets for every SACK received, so the default limit is set
// just above that, and then mostly applicable for (but not limited to) fast
// retransmission scenarios.
int max_burst = 4;
// Maximum Data Retransmit Attempts (per DATA chunk). Set to absl::nullopt for
// no limit.
absl::optional<int> max_retransmissions = 10;
// Max.Init.Retransmits (https://tools.ietf.org/html/rfc4960#section-15). Set
// to absl::nullopt for no limit.
absl::optional<int> max_init_retransmits = 8;
// RFC3758 Partial Reliability Extension
bool enable_partial_reliability = true;
// RFC8260 Stream Schedulers and User Message Interleaving
bool enable_message_interleaving = false;
// If RTO should be added to heartbeat_interval
bool heartbeat_interval_include_rtt = true;
// Disables SCTP packet crc32 verification. For fuzzers only!
bool disable_checksum_verification = false;
// Controls the "zero checksum option" feature, as defined in
// https://www.ietf.org/archive/id/draft-ietf-tsvwg-sctp-zero-checksum-06.html.
// To have this feature enabled, both peers must be configured to use the
// same (defined, not "none") alternate error detection method.
ZeroChecksumAlternateErrorDetectionMethod
zero_checksum_alternate_error_detection_method =
ZeroChecksumAlternateErrorDetectionMethod::None();
};
} // namespace dcsctp
#endif // NET_DCSCTP_PUBLIC_DCSCTP_OPTIONS_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_PUBLIC_DCSCTP_SOCKET_H_
#define NET_DCSCTP_PUBLIC_DCSCTP_SOCKET_H_
#include <cstdint>
#include <memory>
#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 "api/units/timestamp.h"
#include "net/dcsctp/public/dcsctp_handover_state.h"
#include "net/dcsctp/public/dcsctp_message.h"
#include "net/dcsctp/public/dcsctp_options.h"
#include "net/dcsctp/public/packet_observer.h"
#include "net/dcsctp/public/timeout.h"
#include "net/dcsctp/public/types.h"
namespace dcsctp {
// The socket/association state
enum class SocketState {
// The socket is closed.
kClosed,
// The socket has initiated a connection, which is not yet established. Note
// that for incoming connections and for reconnections when the socket is
// already connected, the socket will not transition to this state.
kConnecting,
// The socket is connected, and the connection is established.
kConnected,
// The socket is shutting down, and the connection is not yet closed.
kShuttingDown,
};
// Send options for sending messages
struct SendOptions {
// If the message should be sent with unordered message delivery.
IsUnordered unordered = IsUnordered(false);
// If set, will discard messages that haven't been correctly sent and
// received before the lifetime has expired. This is only available if the
// peer supports Partial Reliability Extension (RFC3758).
absl::optional<DurationMs> lifetime = absl::nullopt;
// If set, limits the number of retransmissions. This is only available
// if the peer supports Partial Reliability Extension (RFC3758).
absl::optional<size_t> max_retransmissions = absl::nullopt;
// If set, will generate lifecycle events for this message. See e.g.
// `DcSctpSocketCallbacks::OnLifecycleMessageFullySent`. This value is decided
// by the client and the library will provide it to all lifecycle callbacks.
LifecycleId lifecycle_id = LifecycleId::NotSet();
};
enum class ErrorKind {
// Indicates that no error has occurred. This will never be the case when
// `OnError` or `OnAborted` is called.
kNoError,
// There have been too many retries or timeouts, and the library has given up.
kTooManyRetries,
// A command was received that is only possible to execute when the socket is
// connected, which it is not.
kNotConnected,
// Parsing of the command or its parameters failed.
kParseFailed,
// Commands are received in the wrong sequence, which indicates a
// synchronisation mismatch between the peers.
kWrongSequence,
// The peer has reported an issue using ERROR or ABORT command.
kPeerReported,
// The peer has performed a protocol violation.
kProtocolViolation,
// The receive or send buffers have been exhausted.
kResourceExhaustion,
// The client has performed an invalid operation.
kUnsupportedOperation,
};
inline constexpr absl::string_view ToString(ErrorKind error) {
switch (error) {
case ErrorKind::kNoError:
return "NO_ERROR";
case ErrorKind::kTooManyRetries:
return "TOO_MANY_RETRIES";
case ErrorKind::kNotConnected:
return "NOT_CONNECTED";
case ErrorKind::kParseFailed:
return "PARSE_FAILED";
case ErrorKind::kWrongSequence:
return "WRONG_SEQUENCE";
case ErrorKind::kPeerReported:
return "PEER_REPORTED";
case ErrorKind::kProtocolViolation:
return "PROTOCOL_VIOLATION";
case ErrorKind::kResourceExhaustion:
return "RESOURCE_EXHAUSTION";
case ErrorKind::kUnsupportedOperation:
return "UNSUPPORTED_OPERATION";
}
}
enum class SendStatus {
// The message was enqueued successfully. As sending the message is done
// asynchronously, this is no guarantee that the message has been actually
// sent.
kSuccess,
// The message was rejected as the payload was empty (which is not allowed in
// SCTP).
kErrorMessageEmpty,
// The message was rejected as the payload was larger than what has been set
// as `DcSctpOptions.max_message_size`.
kErrorMessageTooLarge,
// The message could not be enqueued as the socket is out of resources. This
// mainly indicates that the send queue is full.
kErrorResourceExhaustion,
// The message could not be sent as the socket is shutting down.
kErrorShuttingDown,
};
inline constexpr absl::string_view ToString(SendStatus error) {
switch (error) {
case SendStatus::kSuccess:
return "SUCCESS";
case SendStatus::kErrorMessageEmpty:
return "ERROR_MESSAGE_EMPTY";
case SendStatus::kErrorMessageTooLarge:
return "ERROR_MESSAGE_TOO_LARGE";
case SendStatus::kErrorResourceExhaustion:
return "ERROR_RESOURCE_EXHAUSTION";
case SendStatus::kErrorShuttingDown:
return "ERROR_SHUTTING_DOWN";
}
}
// Return value of ResetStreams.
enum class ResetStreamsStatus {
// If the connection is not yet established, this will be returned.
kNotConnected,
// Indicates that ResetStreams operation has been successfully initiated.
kPerformed,
// Indicates that ResetStreams has failed as it's not supported by the peer.
kNotSupported,
};
inline constexpr absl::string_view ToString(ResetStreamsStatus error) {
switch (error) {
case ResetStreamsStatus::kNotConnected:
return "NOT_CONNECTED";
case ResetStreamsStatus::kPerformed:
return "PERFORMED";
case ResetStreamsStatus::kNotSupported:
return "NOT_SUPPORTED";
}
}
// Return value of DcSctpSocketCallbacks::SendPacketWithStatus.
enum class SendPacketStatus {
// Indicates that the packet was successfully sent. As sending is unreliable,
// there are no guarantees that the packet was actually delivered.
kSuccess,
// The packet was not sent due to a temporary failure, such as the local send
// buffer becoming exhausted. This return value indicates that the socket will
// recover and sending that packet can be retried at a later time.
kTemporaryFailure,
// The packet was not sent due to other reasons.
kError,
};
// Represent known SCTP implementations.
enum class SctpImplementation {
// There is not enough information toto determine any SCTP implementation.
kUnknown,
// This implementation.
kDcsctp,
// https://github.com/sctplab/usrsctp.
kUsrSctp,
// Any other implementation.
kOther,
};
inline constexpr absl::string_view ToString(SctpImplementation implementation) {
switch (implementation) {
case SctpImplementation::kUnknown:
return "unknown";
case SctpImplementation::kDcsctp:
return "dcsctp";
case SctpImplementation::kUsrSctp:
return "usrsctp";
case SctpImplementation::kOther:
return "other";
}
}
// Tracked metrics, which is the return value of GetMetrics. Optional members
// will be unset when they are not yet known.
struct Metrics {
// Transmission stats and metrics.
// Number of packets sent.
size_t tx_packets_count = 0;
// Number of messages requested to be sent.
size_t tx_messages_count = 0;
// Number of packets retransmitted. Since SCTP packets can contain both
// retransmitted DATA chunks and DATA chunks that are transmitted for the
// first time, this represents an upper bound as it's incremented every time a
// packet contains a retransmitted DATA chunk.
size_t rtx_packets_count = 0;
// Total number of bytes retransmitted. This includes the payload and
// DATA/I-DATA headers, but not SCTP packet headers.
uint64_t rtx_bytes_count = 0;
// The current congestion window (cwnd) in bytes, corresponding to spinfo_cwnd
// defined in RFC6458.
size_t cwnd_bytes = 0;
// Smoothed round trip time, corresponding to spinfo_srtt defined in RFC6458.
int srtt_ms = 0;
// Number of data items in the retransmission queue that haven’t been
// acked/nacked yet and are in-flight. Corresponding to sstat_unackdata
// defined in RFC6458. This may be an approximation when there are messages in
// the send queue that haven't been fragmented/packetized yet.
size_t unack_data_count = 0;
// Receive stats and metrics.
// Number of packets received.
size_t rx_packets_count = 0;
// Number of messages received.
size_t rx_messages_count = 0;
// The peer’s last announced receiver window size, corresponding to
// sstat_rwnd defined in RFC6458.
uint32_t peer_rwnd_bytes = 0;
// Returns the detected SCTP implementation of the peer. As this is not
// explicitly signalled during the connection establishment, heuristics is
// used to analyze e.g. the state cookie in the INIT-ACK chunk.
SctpImplementation peer_implementation = SctpImplementation::kUnknown;
// Indicates if RFC8260 User Message Interleaving has been negotiated by both
// peers.
bool uses_message_interleaving = false;
// Indicates if draft-tuexen-tsvwg-sctp-zero-checksum-00 has been negotiated
// by both peers.
bool uses_zero_checksum = false;
// The number of negotiated incoming and outgoing streams, which is configured
// locally as `DcSctpOptions::announced_maximum_incoming_streams` and
// `DcSctpOptions::announced_maximum_outgoing_streams`, and which will be
// signaled by the peer during connection.
uint16_t negotiated_maximum_incoming_streams = 0;
uint16_t negotiated_maximum_outgoing_streams = 0;
};
// Callbacks that the DcSctpSocket will call synchronously to the owning
// client. It is allowed to call back into the library from callbacks that start
// with "On". It has been explicitly documented when it's not allowed to call
// back into this library from within a callback.
//
// Theses callbacks are only synchronously triggered as a result of the client
// calling a public method in `DcSctpSocketInterface`.
class DcSctpSocketCallbacks {
public:
virtual ~DcSctpSocketCallbacks() = default;
// Called when the library wants the packet serialized as `data` to be sent.
//
// TODO(bugs.webrtc.org/12943): This method is deprecated, see
// `SendPacketWithStatus`.
//
// Note that it's NOT ALLOWED to call into this library from within this
// callback.
virtual void SendPacket(rtc::ArrayView<const uint8_t> data) {}
// Called when the library wants the packet serialized as `data` to be sent.
//
// Note that it's NOT ALLOWED to call into this library from within this
// callback.
virtual SendPacketStatus SendPacketWithStatus(
rtc::ArrayView<const uint8_t> data) {
SendPacket(data);
return SendPacketStatus::kSuccess;
}
// Called when the library wants to create a Timeout. The callback must return
// an object that implements that interface.
//
// Low precision tasks are scheduled more efficiently by using leeway to
// reduce Idle Wake Ups and is the preferred precision whenever possible. High
// precision timeouts do not have this leeway, but is still limited by OS
// timer precision. At the time of writing, kLow's additional leeway may be up
// to 17 ms, but please see webrtc::TaskQueueBase::DelayPrecision for
// up-to-date information.
//
// Note that it's NOT ALLOWED to call into this library from within this
// callback.
virtual std::unique_ptr<Timeout> CreateTimeout(
webrtc::TaskQueueBase::DelayPrecision precision) {
// TODO(hbos): When dependencies have migrated to this new signature, make
// this pure virtual and delete the other version.
return CreateTimeout();
}
// TODO(hbos): When dependencies have migrated to the other signature, delete
// this version.
virtual std::unique_ptr<Timeout> CreateTimeout() {
return CreateTimeout(webrtc::TaskQueueBase::DelayPrecision::kLow);
}
// Returns the current time in milliseconds (from any epoch).
//
// TODO(bugs.webrtc.org/15593): This method is deprecated, see `Now`.
//
// Note that it's NOT ALLOWED to call into this library from within this
// callback.
virtual TimeMs TimeMillis() { return TimeMs(0); }
// Returns the current time (from any epoch).
//
// This callback will eventually replace `TimeMillis()`.
//
// Note that it's NOT ALLOWED to call into this library from within this
// callback.
virtual webrtc::Timestamp Now() {
return webrtc::Timestamp::Millis(*TimeMillis());
}
// Called when the library needs a random number uniformly distributed between
// `low` (inclusive) and `high` (exclusive). The random numbers used by the
// library are not used for cryptographic purposes. There are no requirements
// that the random number generator must be secure.
//
// Note that it's NOT ALLOWED to call into this library from within this
// callback.
virtual uint32_t GetRandomInt(uint32_t low, uint32_t high) = 0;
// Triggered when the outgoing message buffer is empty, meaning that there are
// no more queued messages, but there can still be packets in-flight or to be
// retransmitted. (in contrast to SCTP_SENDER_DRY_EVENT).
//
// Note that it's NOT ALLOWED to call into this library from within this
// callback.
ABSL_DEPRECATED("Use OnTotalBufferedAmountLow instead")
virtual void NotifyOutgoingMessageBufferEmpty() {}
// Called when the library has received an SCTP message in full and delivers
// it to the upper layer.
//
// It is allowed to call into this library from within this callback.
virtual void OnMessageReceived(DcSctpMessage message) = 0;
// Triggered when an non-fatal error is reported by either this library or
// from the other peer (by sending an ERROR command). These should be logged,
// but no other action need to be taken as the association is still viable.
//
// It is allowed to call into this library from within this callback.
virtual void OnError(ErrorKind error, absl::string_view message) = 0;
// Triggered when the socket has aborted - either as decided by this socket
// due to e.g. too many retransmission attempts, or by the peer when
// receiving an ABORT command. No other callbacks will be done after this
// callback, unless reconnecting.
//
// It is allowed to call into this library from within this callback.
virtual void OnAborted(ErrorKind error, absl::string_view message) = 0;
// Called when calling `Connect` succeeds, but also for incoming successful
// connection attempts.
//
// It is allowed to call into this library from within this callback.
virtual void OnConnected() = 0;
// Called when the socket is closed in a controlled way. No other
// callbacks will be done after this callback, unless reconnecting.
//
// It is allowed to call into this library from within this callback.
virtual void OnClosed() = 0;
// On connection restarted (by peer). This is just a notification, and the
// association is expected to work fine after this call, but there could have
// been packet loss as a result of restarting the association.
//
// It is allowed to call into this library from within this callback.
virtual void OnConnectionRestarted() = 0;
// Indicates that a stream reset request has failed.
//
// It is allowed to call into this library from within this callback.
virtual void OnStreamsResetFailed(
rtc::ArrayView<const StreamID> outgoing_streams,
absl::string_view reason) = 0;
// Indicates that a stream reset request has been performed.
//
// It is allowed to call into this library from within this callback.
virtual void OnStreamsResetPerformed(
rtc::ArrayView<const StreamID> outgoing_streams) = 0;
// When a peer has reset some of its outgoing streams, this will be called. An
// empty list indicates that all streams have been reset.
//
// It is allowed to call into this library from within this callback.
virtual void OnIncomingStreamsReset(
rtc::ArrayView<const StreamID> incoming_streams) = 0;
// Will be called when the amount of data buffered to be sent falls to or
// below the threshold set when calling `SetBufferedAmountLowThreshold`.
//
// It is allowed to call into this library from within this callback.
virtual void OnBufferedAmountLow(StreamID stream_id) {}
// Will be called when the total amount of data buffered (in the entire send
// buffer, for all streams) falls to or below the threshold specified in
// `DcSctpOptions::total_buffered_amount_low_threshold`.
virtual void OnTotalBufferedAmountLow() {}
// == Lifecycle Events ==
//
// If a `lifecycle_id` is provided as `SendOptions`, lifecycle callbacks will
// be triggered as the message is processed by the library.
//
// The possible transitions are shown in the graph below:
//
// DcSctpSocket::Send ────────────────────────┐
// │ │
// │ │
// v v
// OnLifecycleMessageFullySent ───────> OnLifecycleMessageExpired
// │ │
// │ │
// v v
// OnLifeCycleMessageDelivered ────────────> OnLifecycleEnd
// OnLifecycleMessageFullySent will be called when a message has been fully
// sent, meaning that the last fragment has been produced from the send queue
// and sent on the network. Note that this will trigger at most once per
// message even if the message was retransmitted due to packet loss.
//
// This is a lifecycle event.
//
// Note that it's NOT ALLOWED to call into this library from within this
// callback.
virtual void OnLifecycleMessageFullySent(LifecycleId lifecycle_id) {}
// OnLifecycleMessageExpired will be called when a message has expired. If it
// was expired with data remaining in the send queue that had not been sent
// ever, `maybe_delivered` will be set to false. If `maybe_delivered` is true,
// the message has at least once been sent and may have been correctly
// received by the peer, but it has expired before the receiver managed to
// acknowledge it. This means that if `maybe_delivered` is true, it's unknown
// if the message was lost or was delivered, and if `maybe_delivered` is
// false, it's guaranteed to not be delivered.
//
// It's guaranteed that `OnLifecycleMessageDelivered` is not called if this
// callback has triggered.
//
// This is a lifecycle event.
//
// Note that it's NOT ALLOWED to call into this library from within this
// callback.
virtual void OnLifecycleMessageExpired(LifecycleId lifecycle_id,
bool maybe_delivered) {}
// OnLifecycleMessageDelivered will be called when a non-expired message has
// been acknowledged by the peer as delivered.
//
// Note that this will trigger only when the peer moves its cumulative TSN ack
// beyond this message, and will not fire for messages acked using
// gap-ack-blocks as those are renegable. This means that this may fire a bit
// later than the message was actually first "acked" by the peer, as -
// according to the protocol - those acks may be unacked later by the client.
//
// It's guaranteed that `OnLifecycleMessageExpired` is not called if this
// callback has triggered.
//
// This is a lifecycle event.
//
// Note that it's NOT ALLOWED to call into this library from within this
// callback.
virtual void OnLifecycleMessageDelivered(LifecycleId lifecycle_id) {}
// OnLifecycleEnd will be called when a lifecycle event has reached its end.
// It will be called when processing of a message is complete, no matter how
// it completed. It will be called after all other lifecycle events, if any.
//
// Note that it's possible that this callback triggers without any other
// lifecycle callbacks having been called before in case of errors, such as
// attempting to send an empty message or failing to enqueue a message if the
// send queue is full.
//
// NOTE: When the socket is deallocated, there will be no `OnLifecycleEnd`
// callbacks sent for messages that were enqueued. But as long as the socket
// is alive, `OnLifecycleEnd` callbacks are guaranteed to be sent as messages
// are either expired or successfully acknowledged.
//
// This is a lifecycle event.
//
// Note that it's NOT ALLOWED to call into this library from within this
// callback.
virtual void OnLifecycleEnd(LifecycleId lifecycle_id) {}
};
// The DcSctpSocket implementation implements the following interface.
// This class is thread-compatible.
class DcSctpSocketInterface {
public:
virtual ~DcSctpSocketInterface() = default;
// To be called when an incoming SCTP packet is to be processed.
virtual void ReceivePacket(rtc::ArrayView<const uint8_t> data) = 0;
// To be called when a timeout has expired. The `timeout_id` is provided
// when the timeout was initiated.
virtual void HandleTimeout(TimeoutID timeout_id) = 0;
// Connects the socket. This is an asynchronous operation, and
// `DcSctpSocketCallbacks::OnConnected` will be called on success.
virtual void Connect() = 0;
// Puts this socket to the state in which the original socket was when its
// `DcSctpSocketHandoverState` was captured by `GetHandoverStateAndClose`.
// `RestoreFromState` is allowed only on the closed socket.
// `DcSctpSocketCallbacks::OnConnected` will be called if a connected socket
// state is restored.
// `DcSctpSocketCallbacks::OnError` will be called on error.
virtual void RestoreFromState(const DcSctpSocketHandoverState& state) = 0;
// Gracefully shutdowns the socket and sends all outstanding data. This is an
// asynchronous operation and `DcSctpSocketCallbacks::OnClosed` will be called
// on success.
virtual void Shutdown() = 0;
// Closes the connection non-gracefully. Will send ABORT if the connection is
// not already closed. No callbacks will be made after Close() has returned.
virtual void Close() = 0;
// The socket state.
virtual SocketState state() const = 0;
// The options it was created with.
virtual const DcSctpOptions& options() const = 0;
// Update the options max_message_size.
virtual void SetMaxMessageSize(size_t max_message_size) = 0;
// Sets the priority of an outgoing stream. The initial value, when not set,
// is `DcSctpOptions::default_stream_priority`.
virtual void SetStreamPriority(StreamID stream_id,
StreamPriority priority) = 0;
// Returns the currently set priority for an outgoing stream. The initial
// value, when not set, is `DcSctpOptions::default_stream_priority`.
virtual StreamPriority GetStreamPriority(StreamID stream_id) const = 0;
// Sends the message `message` using the provided send options.
// Sending a message is an asynchronous operation, and the `OnError` callback
// may be invoked to indicate any errors in sending the message.
//
// The association does not have to be established before calling this method.
// If it's called before there is an established association, the message will
// be queued.
virtual SendStatus Send(DcSctpMessage message,
const SendOptions& send_options) = 0;
// Sends the messages `messages` using the provided send options.
// Sending a message is an asynchronous operation, and the `OnError` callback
// may be invoked to indicate any errors in sending the message.
//
// This has identical semantics to Send, except that it may coalesce many
// messages into a single SCTP packet if they would fit.
virtual std::vector<SendStatus> SendMany(
rtc::ArrayView<DcSctpMessage> messages,
const SendOptions& send_options) = 0;
// Resetting streams is an asynchronous operation and the results will
// be notified using `DcSctpSocketCallbacks::OnStreamsResetDone()` on success
// and `DcSctpSocketCallbacks::OnStreamsResetFailed()` on failure. Note that
// only outgoing streams can be reset.
//
// When it's known that the peer has reset its own outgoing streams,
// `DcSctpSocketCallbacks::OnIncomingStreamReset` is called.
//
// Note that resetting a stream will also remove all queued messages on those
// streams, but will ensure that the currently sent message (if any) is fully
// sent before closing the stream.
//
// Resetting streams can only be done on an established association that
// supports stream resetting. Calling this method on e.g. a closed association
// or streams that don't support resetting will not perform any operation.
virtual ResetStreamsStatus ResetStreams(
rtc::ArrayView<const StreamID> outgoing_streams) = 0;
// Returns the number of bytes of data currently queued to be sent on a given
// stream.
virtual size_t buffered_amount(StreamID stream_id) const = 0;
// Returns the number of buffered outgoing bytes that is considered "low" for
// a given stream. See `SetBufferedAmountLowThreshold`.
virtual size_t buffered_amount_low_threshold(StreamID stream_id) const = 0;
// Used to specify the number of bytes of buffered outgoing data that is
// considered "low" for a given stream, which will trigger an
// OnBufferedAmountLow event. The default value is zero (0).
virtual void SetBufferedAmountLowThreshold(StreamID stream_id,
size_t bytes) = 0;
// Retrieves the latest metrics. If the socket is not fully connected,
// `absl::nullopt` will be returned. Note that metrics are not guaranteed to
// be carried over if this socket is handed over by calling
// `GetHandoverStateAndClose`.
virtual absl::optional<Metrics> GetMetrics() const = 0;
// Returns empty bitmask if the socket is in the state in which a snapshot of
// the state can be made by `GetHandoverStateAndClose()`. Return value is
// invalidated by a call to any non-const method.
virtual HandoverReadinessStatus GetHandoverReadiness() const = 0;
// Collects a snapshot of the socket state that can be used to reconstruct
// this socket in another process. On success this socket object is closed
// synchronously and no callbacks will be made after the method has returned.
// The method fails if the socket is not in a state ready for handover.
// nullopt indicates the failure. `DcSctpSocketCallbacks::OnClosed` will be
// called on success.
virtual absl::optional<DcSctpSocketHandoverState>
GetHandoverStateAndClose() = 0;
// Returns the detected SCTP implementation of the peer. As this is not
// explicitly signalled during the connection establishment, heuristics is
// used to analyze e.g. the state cookie in the INIT-ACK chunk.
//
// If this method is called too early (before
// `DcSctpSocketCallbacks::OnConnected` has triggered), this will likely
// return `SctpImplementation::kUnknown`.
ABSL_DEPRECATED("See Metrics::peer_implementation instead")
virtual SctpImplementation peer_implementation() const {
return SctpImplementation::kUnknown;
}
};
} // namespace dcsctp
#endif // NET_DCSCTP_PUBLIC_DCSCTP_SOCKET_H_

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/*
* Copyright 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/public/dcsctp_socket_factory.h"
#include <memory>
#include <utility>
#include "absl/strings/string_view.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/socket/dcsctp_socket.h"
namespace dcsctp {
DcSctpSocketFactory::~DcSctpSocketFactory() = default;
std::unique_ptr<DcSctpSocketInterface> DcSctpSocketFactory::Create(
absl::string_view log_prefix,
DcSctpSocketCallbacks& callbacks,
std::unique_ptr<PacketObserver> packet_observer,
const DcSctpOptions& options) {
return std::make_unique<DcSctpSocket>(log_prefix, callbacks,
std::move(packet_observer), options);
}
} // namespace dcsctp

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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_PUBLIC_DCSCTP_SOCKET_FACTORY_H_
#define NET_DCSCTP_PUBLIC_DCSCTP_SOCKET_FACTORY_H_
#include <memory>
#include "absl/strings/string_view.h"
#include "net/dcsctp/public/dcsctp_options.h"
#include "net/dcsctp/public/dcsctp_socket.h"
#include "net/dcsctp/public/packet_observer.h"
namespace dcsctp {
class DcSctpSocketFactory {
public:
virtual ~DcSctpSocketFactory();
virtual std::unique_ptr<DcSctpSocketInterface> Create(
absl::string_view log_prefix,
DcSctpSocketCallbacks& callbacks,
std::unique_ptr<PacketObserver> packet_observer,
const DcSctpOptions& options);
};
} // namespace dcsctp
#endif // NET_DCSCTP_PUBLIC_DCSCTP_SOCKET_FACTORY_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_PUBLIC_MOCK_DCSCTP_SOCKET_H_
#define NET_DCSCTP_PUBLIC_MOCK_DCSCTP_SOCKET_H_
#include <vector>
#include "net/dcsctp/public/dcsctp_socket.h"
#include "test/gmock.h"
namespace dcsctp {
class MockDcSctpSocket : public DcSctpSocketInterface {
public:
MOCK_METHOD(void,
ReceivePacket,
(rtc::ArrayView<const uint8_t> data),
(override));
MOCK_METHOD(void, HandleTimeout, (TimeoutID timeout_id), (override));
MOCK_METHOD(void, Connect, (), (override));
MOCK_METHOD(void,
RestoreFromState,
(const DcSctpSocketHandoverState&),
(override));
MOCK_METHOD(void, Shutdown, (), (override));
MOCK_METHOD(void, Close, (), (override));
MOCK_METHOD(SocketState, state, (), (const, override));
MOCK_METHOD(const DcSctpOptions&, options, (), (const, override));
MOCK_METHOD(void, SetMaxMessageSize, (size_t max_message_size), (override));
MOCK_METHOD(void,
SetStreamPriority,
(StreamID stream_id, StreamPriority priority),
(override));
MOCK_METHOD(StreamPriority,
GetStreamPriority,
(StreamID stream_id),
(const, override));
MOCK_METHOD(SendStatus,
Send,
(DcSctpMessage message, const SendOptions& send_options),
(override));
MOCK_METHOD(std::vector<SendStatus>,
SendMany,
(rtc::ArrayView<DcSctpMessage> messages,
const SendOptions& send_options),
(override));
MOCK_METHOD(ResetStreamsStatus,
ResetStreams,
(rtc::ArrayView<const StreamID> outgoing_streams),
(override));
MOCK_METHOD(size_t, buffered_amount, (StreamID stream_id), (const, override));
MOCK_METHOD(size_t,
buffered_amount_low_threshold,
(StreamID stream_id),
(const, override));
MOCK_METHOD(void,
SetBufferedAmountLowThreshold,
(StreamID stream_id, size_t bytes),
(override));
MOCK_METHOD(absl::optional<Metrics>, GetMetrics, (), (const, override));
MOCK_METHOD(HandoverReadinessStatus,
GetHandoverReadiness,
(),
(const, override));
MOCK_METHOD(absl::optional<DcSctpSocketHandoverState>,
GetHandoverStateAndClose,
(),
(override));
};
} // namespace dcsctp
#endif // NET_DCSCTP_PUBLIC_MOCK_DCSCTP_SOCKET_H_

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/*
* Copyright (c) 2022 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_PUBLIC_MOCK_DCSCTP_SOCKET_FACTORY_H_
#define NET_DCSCTP_PUBLIC_MOCK_DCSCTP_SOCKET_FACTORY_H_
#include <memory>
#include "net/dcsctp/public/dcsctp_socket_factory.h"
#include "test/gmock.h"
namespace dcsctp {
class MockDcSctpSocketFactory : public DcSctpSocketFactory {
public:
MOCK_METHOD(std::unique_ptr<DcSctpSocketInterface>,
Create,
(absl::string_view log_prefix,
DcSctpSocketCallbacks& callbacks,
std::unique_ptr<PacketObserver> packet_observer,
const DcSctpOptions& options),
(override));
};
} // namespace dcsctp
#endif // NET_DCSCTP_PUBLIC_MOCK_DCSCTP_SOCKET_FACTORY_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_PUBLIC_PACKET_OBSERVER_H_
#define NET_DCSCTP_PUBLIC_PACKET_OBSERVER_H_
#include <stdint.h>
#include "api/array_view.h"
#include "net/dcsctp/public/types.h"
namespace dcsctp {
// A PacketObserver can be attached to a socket and will be called for
// all sent and received packets.
class PacketObserver {
public:
virtual ~PacketObserver() = default;
// Called when a packet is sent, with the current time (in milliseconds) as
// `now`, and the packet payload as `payload`.
virtual void OnSentPacket(TimeMs now,
rtc::ArrayView<const uint8_t> payload) = 0;
// Called when a packet is received, with the current time (in milliseconds)
// as `now`, and the packet payload as `payload`.
virtual void OnReceivedPacket(TimeMs now,
rtc::ArrayView<const uint8_t> payload) = 0;
};
} // namespace dcsctp
#endif // NET_DCSCTP_PUBLIC_PACKET_OBSERVER_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.
*/
#include "net/dcsctp/public/text_pcap_packet_observer.h"
#include "api/array_view.h"
#include "net/dcsctp/public/types.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
namespace dcsctp {
void TextPcapPacketObserver::OnSentPacket(
dcsctp::TimeMs now,
rtc::ArrayView<const uint8_t> payload) {
PrintPacket("O ", name_, now, payload);
}
void TextPcapPacketObserver::OnReceivedPacket(
dcsctp::TimeMs now,
rtc::ArrayView<const uint8_t> payload) {
PrintPacket("I ", name_, now, payload);
}
void TextPcapPacketObserver::PrintPacket(
absl::string_view prefix,
absl::string_view socket_name,
dcsctp::TimeMs now,
rtc::ArrayView<const uint8_t> payload) {
rtc::StringBuilder s;
s << "\n" << prefix;
int64_t remaining = *now % (24 * 60 * 60 * 1000);
int hours = remaining / (60 * 60 * 1000);
remaining = remaining % (60 * 60 * 1000);
int minutes = remaining / (60 * 1000);
remaining = remaining % (60 * 1000);
int seconds = remaining / 1000;
int ms = remaining % 1000;
s.AppendFormat("%02d:%02d:%02d.%03d", hours, minutes, seconds, ms);
s << " 0000";
for (uint8_t byte : payload) {
s.AppendFormat(" %02x", byte);
}
s << " # SCTP_PACKET " << socket_name;
RTC_LOG(LS_VERBOSE) << s.str();
}
} // namespace dcsctp

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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_PUBLIC_TEXT_PCAP_PACKET_OBSERVER_H_
#define NET_DCSCTP_PUBLIC_TEXT_PCAP_PACKET_OBSERVER_H_
#include <string>
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "net/dcsctp/public/packet_observer.h"
#include "net/dcsctp/public/types.h"
namespace dcsctp {
// Print outs all sent and received packets to the logs, at LS_VERBOSE severity.
class TextPcapPacketObserver : public dcsctp::PacketObserver {
public:
explicit TextPcapPacketObserver(absl::string_view name) : name_(name) {}
// Implementation of `dcsctp::PacketObserver`.
void OnSentPacket(dcsctp::TimeMs now,
rtc::ArrayView<const uint8_t> payload) override;
void OnReceivedPacket(dcsctp::TimeMs now,
rtc::ArrayView<const uint8_t> payload) override;
// Prints a packet to the log. Exposed to allow it to be used in compatibility
// tests suites that don't use PacketObserver.
static void PrintPacket(absl::string_view prefix,
absl::string_view socket_name,
dcsctp::TimeMs now,
rtc::ArrayView<const uint8_t> payload);
private:
const std::string name_;
};
} // namespace dcsctp
#endif // NET_DCSCTP_PUBLIC_TEXT_PCAP_PACKET_OBSERVER_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_PUBLIC_TIMEOUT_H_
#define NET_DCSCTP_PUBLIC_TIMEOUT_H_
#include <cstdint>
#include "net/dcsctp/public/types.h"
namespace dcsctp {
// A very simple timeout that can be started and stopped. When started,
// it will be given a unique `timeout_id` which should be provided to
// `DcSctpSocket::HandleTimeout` when it expires.
class Timeout {
public:
virtual ~Timeout() = default;
// Called to start time timeout, with the duration in milliseconds as
// `duration` and with the timeout identifier as `timeout_id`, which - if
// the timeout expires - shall be provided to `DcSctpSocket::HandleTimeout`.
//
// `Start` and `Stop` will always be called in pairs. In other words will
// ´Start` never be called twice, without a call to `Stop` in between.
virtual void Start(DurationMs duration, TimeoutID timeout_id) = 0;
// Called to stop the running timeout.
//
// `Start` and `Stop` will always be called in pairs. In other words will
// ´Start` never be called twice, without a call to `Stop` in between.
//
// `Stop` will always be called prior to releasing this object.
virtual void Stop() = 0;
// Called to restart an already running timeout, with the `duration` and
// `timeout_id` parameters as described in `Start`. This can be overridden by
// the implementation to restart it more efficiently.
virtual void Restart(DurationMs duration, TimeoutID timeout_id) {
Stop();
Start(duration, timeout_id);
}
};
} // namespace dcsctp
#endif // NET_DCSCTP_PUBLIC_TIMEOUT_H_

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/*
* Copyright 2019 The Chromium Authors. All rights reserved.
* 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_PUBLIC_TYPES_H_
#define NET_DCSCTP_PUBLIC_TYPES_H_
#include <cstdint>
#include <limits>
#include "api/units/time_delta.h"
#include "rtc_base/strong_alias.h"
namespace dcsctp {
// Stream Identifier
using StreamID = webrtc::StrongAlias<class StreamIDTag, uint16_t>;
// Payload Protocol Identifier (PPID)
using PPID = webrtc::StrongAlias<class PPIDTag, uint32_t>;
// Timeout Identifier
using TimeoutID = webrtc::StrongAlias<class TimeoutTag, uint64_t>;
// Indicates if a message is allowed to be received out-of-order compared to
// other messages on the same stream.
using IsUnordered = webrtc::StrongAlias<class IsUnorderedTag, bool>;
// Stream priority, where higher values indicate higher priority. The meaning of
// this value and how it's used depends on the stream scheduler.
using StreamPriority = webrtc::StrongAlias<class StreamPriorityTag, uint16_t>;
// Duration, as milliseconds. Overflows after 24 days.
class DurationMs : public webrtc::StrongAlias<class DurationMsTag, int32_t> {
public:
constexpr explicit DurationMs(const UnderlyingType& v)
: webrtc::StrongAlias<class DurationMsTag, int32_t>(v) {}
constexpr explicit DurationMs(webrtc::TimeDelta v)
: webrtc::StrongAlias<class DurationMsTag, int32_t>(
v.IsInfinite() ? InfiniteDuration() : DurationMs(v.ms())) {}
static constexpr DurationMs InfiniteDuration() {
return DurationMs(std::numeric_limits<int32_t>::max());
}
// Convenience methods for working with time.
constexpr DurationMs& operator+=(DurationMs d) {
value_ += d.value_;
return *this;
}
constexpr DurationMs& operator-=(DurationMs d) {
value_ -= d.value_;
return *this;
}
template <typename T>
constexpr DurationMs& operator*=(T factor) {
value_ *= factor;
return *this;
}
constexpr webrtc::TimeDelta ToTimeDelta() const {
return *this == DurationMs::InfiniteDuration()
? webrtc::TimeDelta::PlusInfinity()
: webrtc::TimeDelta::Millis(value_);
}
};
constexpr inline DurationMs operator+(DurationMs lhs, DurationMs rhs) {
return lhs += rhs;
}
constexpr inline DurationMs operator-(DurationMs lhs, DurationMs rhs) {
return lhs -= rhs;
}
template <typename T>
constexpr inline DurationMs operator*(DurationMs lhs, T rhs) {
return lhs *= rhs;
}
template <typename T>
constexpr inline DurationMs operator*(T lhs, DurationMs rhs) {
return rhs *= lhs;
}
constexpr inline int32_t operator/(DurationMs lhs, DurationMs rhs) {
return lhs.value() / rhs.value();
}
// Represents time, in milliseconds since a client-defined epoch.
class TimeMs : public webrtc::StrongAlias<class TimeMsTag, int64_t> {
public:
constexpr explicit TimeMs(const UnderlyingType& v)
: webrtc::StrongAlias<class TimeMsTag, int64_t>(v) {}
// Convenience methods for working with time.
constexpr TimeMs& operator+=(DurationMs d) {
value_ += *d;
return *this;
}
constexpr TimeMs& operator-=(DurationMs d) {
value_ -= *d;
return *this;
}
static constexpr TimeMs InfiniteFuture() {
return TimeMs(std::numeric_limits<int64_t>::max());
}
};
constexpr inline TimeMs operator+(TimeMs lhs, DurationMs rhs) {
return lhs += rhs;
}
constexpr inline TimeMs operator+(DurationMs lhs, TimeMs rhs) {
return rhs += lhs;
}
constexpr inline TimeMs operator-(TimeMs lhs, DurationMs rhs) {
return lhs -= rhs;
}
constexpr inline DurationMs operator-(TimeMs lhs, TimeMs rhs) {
return DurationMs(*lhs - *rhs);
}
// The maximum number of times the socket should attempt to retransmit a
// message which fails the first time in unreliable mode.
class MaxRetransmits
: public webrtc::StrongAlias<class MaxRetransmitsTag, uint16_t> {
public:
constexpr explicit MaxRetransmits(const UnderlyingType& v)
: webrtc::StrongAlias<class MaxRetransmitsTag, uint16_t>(v) {}
// There should be no limit - the message should be sent reliably.
static constexpr MaxRetransmits NoLimit() {
return MaxRetransmits(std::numeric_limits<uint16_t>::max());
}
};
// An identifier that can be set on sent messages, and picked by the sending
// client. If different from `::NotSet()`, lifecycle events will be generated,
// and eventually `DcSctpSocketCallbacks::OnLifecycleEnd` will be called to
// indicate that the lifecycle isn't tracked any longer. The value zero (0) is
// not a valid lifecycle identifier, and will be interpreted as not having it
// set.
class LifecycleId : public webrtc::StrongAlias<class LifecycleIdTag, uint64_t> {
public:
constexpr explicit LifecycleId(const UnderlyingType& v)
: webrtc::StrongAlias<class LifecycleIdTag, uint64_t>(v) {}
constexpr bool IsSet() const { return value_ != 0; }
static constexpr LifecycleId NotSet() { return LifecycleId(0); }
};
// To enable zero checksum feature, both peers must agree on which alternate
// error detection method that is used. See
// https://www.ietf.org/archive/id/draft-ietf-tsvwg-sctp-zero-checksum-06.html.
class ZeroChecksumAlternateErrorDetectionMethod
: public webrtc::StrongAlias<
class ZeroChecksumAlternateErrorDetectionMethodTag,
uint32_t> {
public:
constexpr explicit ZeroChecksumAlternateErrorDetectionMethod(
const UnderlyingType& v)
: webrtc::StrongAlias<class ZeroChecksumAlternateErrorDetectionMethodTag,
uint32_t>(v) {}
static constexpr ZeroChecksumAlternateErrorDetectionMethod None() {
return ZeroChecksumAlternateErrorDetectionMethod(0);
}
static constexpr ZeroChecksumAlternateErrorDetectionMethod LowerLayerDtls() {
return ZeroChecksumAlternateErrorDetectionMethod(1);
}
};
} // namespace dcsctp
#endif // NET_DCSCTP_PUBLIC_TYPES_H_