286 lines
12 KiB
C
286 lines
12 KiB
C
|
/*
|
||
|
|
* Copyright 2020 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 PC_SCTP_DATA_CHANNEL_H_
|
||
|
|
#define PC_SCTP_DATA_CHANNEL_H_
|
||
|
|
|
||
|
|
#include <memory>
|
||
|
|
#include <set>
|
||
|
|
#include <string>
|
||
|
|
|
||
|
|
#include "api/data_channel_interface.h"
|
||
|
|
#include "api/priority.h"
|
||
|
|
#include "api/scoped_refptr.h"
|
||
|
|
#include "api/transport/data_channel_transport_interface.h"
|
||
|
|
#include "media/base/media_channel.h"
|
||
|
|
#include "pc/data_channel_utils.h"
|
||
|
|
#include "rtc_base/async_invoker.h"
|
||
|
|
#include "rtc_base/ssl_stream_adapter.h" // For SSLRole
|
||
|
|
#include "rtc_base/third_party/sigslot/sigslot.h"
|
||
|
|
|
||
|
|
namespace webrtc {
|
||
|
|
|
||
|
|
class SctpDataChannel;
|
||
|
|
|
||
|
|
// TODO(deadbeef): Get rid of this and have SctpDataChannel depend on
|
||
|
|
// SctpTransportInternal (pure virtual SctpTransport interface) instead.
|
||
|
|
class SctpDataChannelProviderInterface {
|
||
|
|
public:
|
||
|
|
// Sends the data to the transport.
|
||
|
|
virtual bool SendData(const cricket::SendDataParams& params,
|
||
|
|
const rtc::CopyOnWriteBuffer& payload,
|
||
|
|
cricket::SendDataResult* result) = 0;
|
||
|
|
// Connects to the transport signals.
|
||
|
|
virtual bool ConnectDataChannel(SctpDataChannel* data_channel) = 0;
|
||
|
|
// Disconnects from the transport signals.
|
||
|
|
virtual void DisconnectDataChannel(SctpDataChannel* data_channel) = 0;
|
||
|
|
// Adds the data channel SID to the transport for SCTP.
|
||
|
|
virtual void AddSctpDataStream(int sid) = 0;
|
||
|
|
// Begins the closing procedure by sending an outgoing stream reset. Still
|
||
|
|
// need to wait for callbacks to tell when this completes.
|
||
|
|
virtual void RemoveSctpDataStream(int sid) = 0;
|
||
|
|
// Returns true if the transport channel is ready to send data.
|
||
|
|
virtual bool ReadyToSendData() const = 0;
|
||
|
|
|
||
|
|
protected:
|
||
|
|
virtual ~SctpDataChannelProviderInterface() {}
|
||
|
|
};
|
||
|
|
|
||
|
|
// TODO(tommi): Change to not inherit from DataChannelInit but to have it as
|
||
|
|
// a const member. Block access to the 'id' member since it cannot be const.
|
||
|
|
struct InternalDataChannelInit : public DataChannelInit {
|
||
|
|
enum OpenHandshakeRole { kOpener, kAcker, kNone };
|
||
|
|
// The default role is kOpener because the default |negotiated| is false.
|
||
|
|
InternalDataChannelInit() : open_handshake_role(kOpener) {}
|
||
|
|
explicit InternalDataChannelInit(const DataChannelInit& base);
|
||
|
|
OpenHandshakeRole open_handshake_role;
|
||
|
|
};
|
||
|
|
|
||
|
|
// Helper class to allocate unique IDs for SCTP DataChannels.
|
||
|
|
class SctpSidAllocator {
|
||
|
|
public:
|
||
|
|
// Gets the first unused odd/even id based on the DTLS role. If |role| is
|
||
|
|
// SSL_CLIENT, the allocated id starts from 0 and takes even numbers;
|
||
|
|
// otherwise, the id starts from 1 and takes odd numbers.
|
||
|
|
// Returns false if no ID can be allocated.
|
||
|
|
bool AllocateSid(rtc::SSLRole role, int* sid);
|
||
|
|
|
||
|
|
// Attempts to reserve a specific sid. Returns false if it's unavailable.
|
||
|
|
bool ReserveSid(int sid);
|
||
|
|
|
||
|
|
// Indicates that |sid| isn't in use any more, and is thus available again.
|
||
|
|
void ReleaseSid(int sid);
|
||
|
|
|
||
|
|
private:
|
||
|
|
// Checks if |sid| is available to be assigned to a new SCTP data channel.
|
||
|
|
bool IsSidAvailable(int sid) const;
|
||
|
|
|
||
|
|
std::set<int> used_sids_;
|
||
|
|
};
|
||
|
|
|
||
|
|
// SctpDataChannel is an implementation of the DataChannelInterface based on
|
||
|
|
// SctpTransport. It provides an implementation of unreliable or
|
||
|
|
// reliabledata channels.
|
||
|
|
|
||
|
|
// DataChannel states:
|
||
|
|
// kConnecting: The channel has been created the transport might not yet be
|
||
|
|
// ready.
|
||
|
|
// kOpen: The open handshake has been performed (if relevant) and the data
|
||
|
|
// channel is able to send messages.
|
||
|
|
// kClosing: DataChannelInterface::Close has been called, or the remote side
|
||
|
|
// initiated the closing procedure, but the closing procedure has not
|
||
|
|
// yet finished.
|
||
|
|
// kClosed: The closing handshake is finished (possibly initiated from this,
|
||
|
|
// side, possibly from the peer).
|
||
|
|
//
|
||
|
|
// How the closing procedure works for SCTP:
|
||
|
|
// 1. Alice calls Close(), state changes to kClosing.
|
||
|
|
// 2. Alice finishes sending any queued data.
|
||
|
|
// 3. Alice calls RemoveSctpDataStream, sends outgoing stream reset.
|
||
|
|
// 4. Bob receives incoming stream reset; OnClosingProcedureStartedRemotely
|
||
|
|
// called.
|
||
|
|
// 5. Bob sends outgoing stream reset.
|
||
|
|
// 6. Alice receives incoming reset, Bob receives acknowledgement. Both receive
|
||
|
|
// OnClosingProcedureComplete callback and transition to kClosed.
|
||
|
|
class SctpDataChannel : public DataChannelInterface,
|
||
|
|
public sigslot::has_slots<> {
|
||
|
|
public:
|
||
|
|
static rtc::scoped_refptr<SctpDataChannel> Create(
|
||
|
|
SctpDataChannelProviderInterface* provider,
|
||
|
|
const std::string& label,
|
||
|
|
const InternalDataChannelInit& config,
|
||
|
|
rtc::Thread* signaling_thread,
|
||
|
|
rtc::Thread* network_thread);
|
||
|
|
|
||
|
|
// Instantiates an API proxy for a SctpDataChannel instance that will be
|
||
|
|
// handed out to external callers.
|
||
|
|
static rtc::scoped_refptr<DataChannelInterface> CreateProxy(
|
||
|
|
rtc::scoped_refptr<SctpDataChannel> channel);
|
||
|
|
|
||
|
|
void RegisterObserver(DataChannelObserver* observer) override;
|
||
|
|
void UnregisterObserver() override;
|
||
|
|
|
||
|
|
std::string label() const override { return label_; }
|
||
|
|
bool reliable() const override;
|
||
|
|
bool ordered() const override { return config_.ordered; }
|
||
|
|
// Backwards compatible accessors
|
||
|
|
uint16_t maxRetransmitTime() const override {
|
||
|
|
return config_.maxRetransmitTime ? *config_.maxRetransmitTime
|
||
|
|
: static_cast<uint16_t>(-1);
|
||
|
|
}
|
||
|
|
uint16_t maxRetransmits() const override {
|
||
|
|
return config_.maxRetransmits ? *config_.maxRetransmits
|
||
|
|
: static_cast<uint16_t>(-1);
|
||
|
|
}
|
||
|
|
absl::optional<int> maxPacketLifeTime() const override {
|
||
|
|
return config_.maxRetransmitTime;
|
||
|
|
}
|
||
|
|
absl::optional<int> maxRetransmitsOpt() const override {
|
||
|
|
return config_.maxRetransmits;
|
||
|
|
}
|
||
|
|
std::string protocol() const override { return config_.protocol; }
|
||
|
|
bool negotiated() const override { return config_.negotiated; }
|
||
|
|
int id() const override { return config_.id; }
|
||
|
|
Priority priority() const override {
|
||
|
|
return config_.priority ? *config_.priority : Priority::kLow;
|
||
|
|
}
|
||
|
|
|
||
|
|
virtual int internal_id() const { return internal_id_; }
|
||
|
|
|
||
|
|
uint64_t buffered_amount() const override;
|
||
|
|
void Close() override;
|
||
|
|
DataState state() const override;
|
||
|
|
RTCError error() const override;
|
||
|
|
uint32_t messages_sent() const override;
|
||
|
|
uint64_t bytes_sent() const override;
|
||
|
|
uint32_t messages_received() const override;
|
||
|
|
uint64_t bytes_received() const override;
|
||
|
|
bool Send(const DataBuffer& buffer) override;
|
||
|
|
|
||
|
|
// Close immediately, ignoring any queued data or closing procedure.
|
||
|
|
// This is called when the underlying SctpTransport is being destroyed.
|
||
|
|
// It is also called by the PeerConnection if SCTP ID assignment fails.
|
||
|
|
void CloseAbruptlyWithError(RTCError error);
|
||
|
|
// Specializations of CloseAbruptlyWithError
|
||
|
|
void CloseAbruptlyWithDataChannelFailure(const std::string& message);
|
||
|
|
void CloseAbruptlyWithSctpCauseCode(const std::string& message,
|
||
|
|
uint16_t cause_code);
|
||
|
|
|
||
|
|
// Slots for provider to connect signals to.
|
||
|
|
//
|
||
|
|
// TODO(deadbeef): Make these private once we're hooking up signals ourselves,
|
||
|
|
// instead of relying on SctpDataChannelProviderInterface.
|
||
|
|
|
||
|
|
// Called when the SctpTransport's ready to use. That can happen when we've
|
||
|
|
// finished negotiation, or if the channel was created after negotiation has
|
||
|
|
// already finished.
|
||
|
|
void OnTransportReady(bool writable);
|
||
|
|
|
||
|
|
void OnDataReceived(const cricket::ReceiveDataParams& params,
|
||
|
|
const rtc::CopyOnWriteBuffer& payload);
|
||
|
|
|
||
|
|
// Sets the SCTP sid and adds to transport layer if not set yet. Should only
|
||
|
|
// be called once.
|
||
|
|
void SetSctpSid(int sid);
|
||
|
|
// The remote side started the closing procedure by resetting its outgoing
|
||
|
|
// stream (our incoming stream). Sets state to kClosing.
|
||
|
|
void OnClosingProcedureStartedRemotely(int sid);
|
||
|
|
// The closing procedure is complete; both incoming and outgoing stream
|
||
|
|
// resets are done and the channel can transition to kClosed. Called
|
||
|
|
// asynchronously after RemoveSctpDataStream.
|
||
|
|
void OnClosingProcedureComplete(int sid);
|
||
|
|
// Called when the transport channel is created.
|
||
|
|
// Only needs to be called for SCTP data channels.
|
||
|
|
void OnTransportChannelCreated();
|
||
|
|
// Called when the transport channel is unusable.
|
||
|
|
// This method makes sure the DataChannel is disconnected and changes state
|
||
|
|
// to kClosed.
|
||
|
|
void OnTransportChannelClosed();
|
||
|
|
|
||
|
|
DataChannelStats GetStats() const;
|
||
|
|
|
||
|
|
// Emitted when state transitions to kOpen.
|
||
|
|
sigslot::signal1<DataChannelInterface*> SignalOpened;
|
||
|
|
// Emitted when state transitions to kClosed.
|
||
|
|
// This signal can be used to tell when the channel's sid is free.
|
||
|
|
sigslot::signal1<DataChannelInterface*> SignalClosed;
|
||
|
|
|
||
|
|
// Reset the allocator for internal ID values for testing, so that
|
||
|
|
// the internal IDs generated are predictable. Test only.
|
||
|
|
static void ResetInternalIdAllocatorForTesting(int new_value);
|
||
|
|
|
||
|
|
protected:
|
||
|
|
SctpDataChannel(const InternalDataChannelInit& config,
|
||
|
|
SctpDataChannelProviderInterface* client,
|
||
|
|
const std::string& label,
|
||
|
|
rtc::Thread* signaling_thread,
|
||
|
|
rtc::Thread* network_thread);
|
||
|
|
~SctpDataChannel() override;
|
||
|
|
|
||
|
|
private:
|
||
|
|
// The OPEN(_ACK) signaling state.
|
||
|
|
enum HandshakeState {
|
||
|
|
kHandshakeInit,
|
||
|
|
kHandshakeShouldSendOpen,
|
||
|
|
kHandshakeShouldSendAck,
|
||
|
|
kHandshakeWaitingForAck,
|
||
|
|
kHandshakeReady
|
||
|
|
};
|
||
|
|
|
||
|
|
bool Init();
|
||
|
|
void UpdateState();
|
||
|
|
void SetState(DataState state);
|
||
|
|
void DisconnectFromProvider();
|
||
|
|
|
||
|
|
void DeliverQueuedReceivedData();
|
||
|
|
|
||
|
|
void SendQueuedDataMessages();
|
||
|
|
bool SendDataMessage(const DataBuffer& buffer, bool queue_if_blocked);
|
||
|
|
bool QueueSendDataMessage(const DataBuffer& buffer);
|
||
|
|
|
||
|
|
void SendQueuedControlMessages();
|
||
|
|
void QueueControlMessage(const rtc::CopyOnWriteBuffer& buffer);
|
||
|
|
bool SendControlMessage(const rtc::CopyOnWriteBuffer& buffer);
|
||
|
|
|
||
|
|
rtc::Thread* const signaling_thread_;
|
||
|
|
rtc::Thread* const network_thread_;
|
||
|
|
const int internal_id_;
|
||
|
|
const std::string label_;
|
||
|
|
const InternalDataChannelInit config_;
|
||
|
|
DataChannelObserver* observer_ RTC_GUARDED_BY(signaling_thread_) = nullptr;
|
||
|
|
DataState state_ RTC_GUARDED_BY(signaling_thread_) = kConnecting;
|
||
|
|
RTCError error_ RTC_GUARDED_BY(signaling_thread_);
|
||
|
|
uint32_t messages_sent_ RTC_GUARDED_BY(signaling_thread_) = 0;
|
||
|
|
uint64_t bytes_sent_ RTC_GUARDED_BY(signaling_thread_) = 0;
|
||
|
|
uint32_t messages_received_ RTC_GUARDED_BY(signaling_thread_) = 0;
|
||
|
|
uint64_t bytes_received_ RTC_GUARDED_BY(signaling_thread_) = 0;
|
||
|
|
// Number of bytes of data that have been queued using Send(). Increased
|
||
|
|
// before each transport send and decreased after each successful send.
|
||
|
|
uint64_t buffered_amount_ RTC_GUARDED_BY(signaling_thread_) = 0;
|
||
|
|
SctpDataChannelProviderInterface* const provider_
|
||
|
|
RTC_GUARDED_BY(signaling_thread_);
|
||
|
|
HandshakeState handshake_state_ RTC_GUARDED_BY(signaling_thread_) =
|
||
|
|
kHandshakeInit;
|
||
|
|
bool connected_to_provider_ RTC_GUARDED_BY(signaling_thread_) = false;
|
||
|
|
bool writable_ RTC_GUARDED_BY(signaling_thread_) = false;
|
||
|
|
// Did we already start the graceful SCTP closing procedure?
|
||
|
|
bool started_closing_procedure_ RTC_GUARDED_BY(signaling_thread_) = false;
|
||
|
|
// Control messages that always have to get sent out before any queued
|
||
|
|
// data.
|
||
|
|
PacketQueue queued_control_data_ RTC_GUARDED_BY(signaling_thread_);
|
||
|
|
PacketQueue queued_received_data_ RTC_GUARDED_BY(signaling_thread_);
|
||
|
|
PacketQueue queued_send_data_ RTC_GUARDED_BY(signaling_thread_);
|
||
|
|
rtc::AsyncInvoker invoker_ RTC_GUARDED_BY(signaling_thread_);
|
||
|
|
};
|
||
|
|
|
||
|
|
} // namespace webrtc
|
||
|
|
|
||
|
|
#endif // PC_SCTP_DATA_CHANNEL_H_
|