blob: 06988fd1560f08d7fe7ad2603199fe2703654cc3 [file] [log] [blame]
* Copyright (c) 2012 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 <errno.h>
#include <cstdint>
#include <map>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "rtc_base/buffer.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread.h"
// For SendDataParams/ReceiveDataParams.
#include "media/base/media_channel.h"
#include "media/sctp/sctp_transport_internal.h"
// Defined by "usrsctplib/usrsctp.h"
struct sockaddr_conn;
struct sctp_assoc_change;
struct sctp_rcvinfo;
struct sctp_stream_reset_event;
struct sctp_sendv_spa;
// Defined by <sys/socket.h>
struct socket;
namespace cricket {
// Holds data to be passed on to a transport.
struct SctpInboundPacket;
// From transport calls, data flows like this:
// [network thread (although it can in princple be another thread)]
// 1. SctpTransport::SendData(data)
// 2. usrsctp_sendv(data)
// [network thread returns; sctp thread then calls the following]
// 3. OnSctpOutboundPacket(wrapped_data)
// [sctp thread returns having async invoked on the network thread]
// 4. SctpTransport::OnPacketFromSctpToNetwork(wrapped_data)
// 5. DtlsTransport::SendPacket(wrapped_data)
// 6. ... across network ... a packet is sent back ...
// 7. SctpTransport::OnPacketReceived(wrapped_data)
// 8. usrsctp_conninput(wrapped_data)
// [network thread returns; sctp thread then calls the following]
// 9. OnSctpInboundData(data)
// 10. SctpTransport::OnDataFromSctpToTransport(data)
// [sctp thread returns having async invoked on the network thread]
// 11. SctpTransport::OnDataFromSctpToTransport(data)
// 12. SctpTransport::SignalDataReceived(data)
// [from the same thread, methods registered/connected to
// SctpTransport are called with the recieved data]
class UsrsctpTransport : public SctpTransportInternal,
public sigslot::has_slots<> {
// `network_thread` is where packets will be processed and callbacks from
// this transport will be posted, and is the only thread on which public
// methods can be called.
// `transport` is not required (can be null).
UsrsctpTransport(rtc::Thread* network_thread,
rtc::PacketTransportInternal* transport);
~UsrsctpTransport() override;
// SctpTransportInternal overrides (see sctptransportinternal.h for comments).
void SetDtlsTransport(rtc::PacketTransportInternal* transport) override;
bool Start(int local_port, int remote_port, int max_message_size) override;
bool OpenStream(int sid) override;
bool ResetStream(int sid) override;
bool SendData(int sid,
const webrtc::SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result = nullptr) override;
bool ReadyToSendData() override;
int max_message_size() const override { return max_message_size_; }
absl::optional<int> max_outbound_streams() const override {
return max_outbound_streams_;
absl::optional<int> max_inbound_streams() const override {
return max_inbound_streams_;
void set_debug_name_for_testing(const char* debug_name) override {
debug_name_ = debug_name;
void InjectDataOrNotificationFromSctpForTesting(const void* data,
size_t length,
struct sctp_rcvinfo rcv,
int flags);
// Exposed to allow Post call from c-callbacks.
// TODO(deadbeef): Remove this or at least make it return a const pointer.
rtc::Thread* network_thread() const { return network_thread_; }
// A message to be sent by the sctp library. This class is used to track the
// progress of writing a single message to the sctp library in the presence of
// partial writes. In this case, the Advance() function is provided in order
// to advance over what has already been accepted by the sctp library and
// avoid copying the remaining partial message buffer.
class OutgoingMessage {
OutgoingMessage(const rtc::CopyOnWriteBuffer& buffer,
int sid,
const webrtc::SendDataParams& send_params)
: buffer_(buffer), sid_(sid), send_params_(send_params) {}
// Advances the buffer by the incremented amount. Must not advance further
// than the current data size.
void Advance(size_t increment) {
RTC_DCHECK_LE(increment + offset_, buffer_.size());
offset_ += increment;
size_t size() const { return buffer_.size() - offset_; }
const void* data() const { return + offset_; }
int sid() const { return sid_; }
webrtc::SendDataParams send_params() const { return send_params_; }
const rtc::CopyOnWriteBuffer buffer_;
int sid_;
const webrtc::SendDataParams send_params_;
size_t offset_ = 0;
void ConnectTransportSignals();
void DisconnectTransportSignals();
// Creates the socket and connects.
bool Connect();
// Returns false when opening the socket failed.
bool OpenSctpSocket();
// Helpet method to set socket options.
bool ConfigureSctpSocket();
// Sets |sock_ |to nullptr.
void CloseSctpSocket();
// Sends a SCTP_RESET_STREAM for all streams in closing_ssids_.
bool SendQueuedStreamResets();
// Sets the "ready to send" flag and fires signal if needed.
void SetReadyToSendData();
// Sends the outgoing buffered message that was only partially accepted by the
// sctp lib because it did not have enough space. Returns true if the entire
// buffered message was accepted by the sctp lib.
bool SendBufferedMessage();
// Tries to send the `payload` on the usrsctp lib. The message will be
// advanced by the amount that was sent.
SendDataResult SendMessageInternal(OutgoingMessage* message);
// Callbacks from DTLS transport.
void OnWritableState(rtc::PacketTransportInternal* transport);
virtual void OnPacketRead(rtc::PacketTransportInternal* transport,
const char* data,
size_t len,
const int64_t& packet_time_us,
int flags);
void OnClosed(rtc::PacketTransportInternal* transport);
// Methods related to usrsctp callbacks.
void OnSendThresholdCallback();
sockaddr_conn GetSctpSockAddr(int port);
// Called using `invoker_` to send packet on the network.
void OnPacketFromSctpToNetwork(const rtc::CopyOnWriteBuffer& buffer);
// Called on the network thread.
// Flags are standard socket API flags (RFC 6458).
void OnDataOrNotificationFromSctp(const void* data,
size_t length,
struct sctp_rcvinfo rcv,
int flags);
// Called using `invoker_` to decide what to do with the data.
void OnDataFromSctpToTransport(const ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& buffer);
// Called using `invoker_` to decide what to do with the notification.
void OnNotificationFromSctp(const rtc::CopyOnWriteBuffer& buffer);
void OnNotificationAssocChange(const sctp_assoc_change& change);
void OnStreamResetEvent(const struct sctp_stream_reset_event* evt);
// Responsible for marshalling incoming data to the transports listeners, and
// outgoing data to the network interface.
rtc::Thread* network_thread_;
// Helps pass inbound/outbound packets asynchronously to the network thread.
webrtc::ScopedTaskSafety task_safety_;
// Underlying DTLS transport.
rtc::PacketTransportInternal* transport_ = nullptr;
// Track the data received from usrsctp between callbacks until the EOR bit
// arrives.
rtc::CopyOnWriteBuffer partial_incoming_message_;
ReceiveDataParams partial_params_;
int partial_flags_;
// A message that was attempted to be sent, but was only partially accepted by
// usrsctp lib with usrsctp_sendv() because it cannot buffer the full message.
// This occurs because we explicitly set the EOR bit when sending, so
// usrsctp_sendv() is not atomic.
absl::optional<OutgoingMessage> partial_outgoing_message_;
bool was_ever_writable_ = false;
int local_port_ = kSctpDefaultPort;
int remote_port_ = kSctpDefaultPort;
int max_message_size_ = kSctpSendBufferSize;
struct socket* sock_ = nullptr; // The socket created by usrsctp_socket(...).
// Has Start been called? Don't create SCTP socket until it has.
bool started_ = false;
// Are we ready to queue data (SCTP socket created, and not blocked due to
// congestion control)? Different than `transport_`'s "ready to send".
bool ready_to_send_data_ = false;
// Used to keep track of the status of each stream (or rather, each pair of
// incoming/outgoing streams with matching IDs). It's specifically used to
// keep track of the status of resets, but more information could be put here
// later.
// See datachannel.h for a summary of the closing procedure.
struct StreamStatus {
// Closure initiated by application via ResetStream? Note that
// this may be true while outgoing_reset_initiated is false if the outgoing
// reset needed to be queued.
bool closure_initiated = false;
// Whether we've initiated the outgoing stream reset via
bool outgoing_reset_initiated = false;
// Whether usrsctp has indicated that the incoming/outgoing streams have
// been reset. It's expected that the peer will reset its outgoing stream
// (our incoming stream) after receiving the reset for our outgoing stream,
// though older versions of chromium won't do this. See
// for context.
bool outgoing_reset_complete = false;
bool incoming_reset_complete = false;
// Some helper methods to improve code readability.
bool is_open() const {
return !closure_initiated && !incoming_reset_complete &&
// We need to send an outgoing reset if the application has closed the data
// channel, or if we received a reset of the incoming stream from the
// remote endpoint, indicating the data channel was closed remotely.
bool need_outgoing_reset() const {
return (incoming_reset_complete || closure_initiated) &&
bool reset_complete() const {
return outgoing_reset_complete && incoming_reset_complete;
// Entries should only be removed from this map if `reset_complete` is
// true.
std::map<uint32_t, StreamStatus> stream_status_by_sid_;
// A static human-readable name for debugging messages.
const char* debug_name_ = "UsrsctpTransport";
// Hides usrsctp interactions from this header file.
class UsrSctpWrapper;
// Number of channels negotiated. Not set before negotiation completes.
absl::optional<int> max_outbound_streams_;
absl::optional<int> max_inbound_streams_;
// Used for associating this transport with the underlying sctp socket in
// various callbacks.
uintptr_t id_ = 0;
friend class UsrsctpTransportMap;
class UsrsctpTransportMap;
} // namespace cricket