2020-08-14 16:58:22 +00:00
|
|
|
/*
|
|
|
|
* 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.
|
|
|
|
*/
|
|
|
|
|
|
|
|
#ifndef MEDIA_SCTP_SCTP_TRANSPORT_H_
|
|
|
|
#define MEDIA_SCTP_SCTP_TRANSPORT_H_
|
|
|
|
|
|
|
|
#include <errno.h>
|
|
|
|
|
|
|
|
#include <cstdint>
|
|
|
|
#include <map>
|
|
|
|
#include <memory>
|
|
|
|
#include <set>
|
|
|
|
#include <string>
|
|
|
|
#include <vector>
|
|
|
|
|
|
|
|
#include "absl/types/optional.h"
|
2020-12-23 07:48:30 +00:00
|
|
|
#include "api/transport/sctp_transport_factory_interface.h"
|
2020-08-14 16:58:22 +00:00
|
|
|
#include "rtc_base/async_invoker.h"
|
|
|
|
#include "rtc_base/buffer.h"
|
|
|
|
#include "rtc_base/constructor_magic.h"
|
|
|
|
#include "rtc_base/copy_on_write_buffer.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 SctpTransport : public SctpTransportInternal,
|
|
|
|
public sigslot::has_slots<> {
|
|
|
|
public:
|
|
|
|
// |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).
|
|
|
|
SctpTransport(rtc::Thread* network_thread,
|
|
|
|
rtc::PacketTransportInternal* transport);
|
|
|
|
~SctpTransport() 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(const 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;
|
|
|
|
}
|
|
|
|
int InjectDataOrNotificationFromSctpForTesting(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_; }
|
|
|
|
|
|
|
|
private:
|
|
|
|
// 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 {
|
|
|
|
public:
|
|
|
|
OutgoingMessage(const rtc::CopyOnWriteBuffer& buffer,
|
|
|
|
const SendDataParams& send_params)
|
|
|
|
: buffer_(buffer), 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 buffer_.data() + offset_; }
|
|
|
|
|
|
|
|
SendDataParams send_params() const { return send_params_; }
|
|
|
|
|
|
|
|
private:
|
|
|
|
const rtc::CopyOnWriteBuffer buffer_;
|
|
|
|
const 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 SCTP thread.
|
|
|
|
// Flags are standard socket API flags (RFC 6458).
|
|
|
|
int OnDataOrNotificationFromSctp(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.
|
|
|
|
rtc::AsyncInvoker invoker_;
|
|
|
|
// 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
|
|
|
|
// SCTP_RESET_STREAMS.
|
|
|
|
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 crbug.com/559394
|
|
|
|
// 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 &&
|
|
|
|
!outgoing_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) &&
|
|
|
|
!outgoing_reset_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_ = "SctpTransport";
|
|
|
|
// 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;
|
|
|
|
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(SctpTransport);
|
|
|
|
};
|
|
|
|
|
2020-12-23 07:48:30 +00:00
|
|
|
class SctpTransportFactory : public webrtc::SctpTransportFactoryInterface {
|
2020-08-14 16:58:22 +00:00
|
|
|
public:
|
|
|
|
explicit SctpTransportFactory(rtc::Thread* network_thread)
|
|
|
|
: network_thread_(network_thread) {}
|
|
|
|
|
|
|
|
std::unique_ptr<SctpTransportInternal> CreateSctpTransport(
|
|
|
|
rtc::PacketTransportInternal* transport) override {
|
|
|
|
return std::unique_ptr<SctpTransportInternal>(
|
|
|
|
new SctpTransport(network_thread_, transport));
|
|
|
|
}
|
|
|
|
|
|
|
|
private:
|
|
|
|
rtc::Thread* network_thread_;
|
|
|
|
};
|
|
|
|
|
|
|
|
} // namespace cricket
|
|
|
|
|
|
|
|
#endif // MEDIA_SCTP_SCTP_TRANSPORT_H_
|