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/*
* Copyright 2019 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 API_SCTP_TRANSPORT_INTERFACE_H_
#define API_SCTP_TRANSPORT_INTERFACE_H_
#include <optional>
#include "api/dtls_transport_interface.h"
#include "api/ref_count.h"
#include "api/scoped_refptr.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
// States of a SCTP transport, corresponding to the JS API specification.
// http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate
enum class SctpTransportState {
kNew, // Has not started negotiating yet. Non-standard state.
kConnecting, // In the process of negotiating an association.
kConnected, // Completed negotiation of an association.
kClosed, // Closed by local or remote party.
kNumValues
};
// This object gives snapshot information about the changeable state of a
// SctpTransport.
// It reflects the readonly attributes of the object in the specification.
// http://w3c.github.io/webrtc-pc/#rtcsctptransport-interface
class RTC_EXPORT SctpTransportInformation {
public:
SctpTransportInformation() = default;
SctpTransportInformation(const SctpTransportInformation&) = default;
explicit SctpTransportInformation(SctpTransportState state);
SctpTransportInformation(
SctpTransportState state,
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport,
std::optional<double> max_message_size,
std::optional<int> max_channels);
~SctpTransportInformation();
// The DTLS transport that supports this SCTP transport.
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const {
return dtls_transport_;
}
SctpTransportState state() const { return state_; }
std::optional<double> MaxMessageSize() const { return max_message_size_; }
std::optional<int> MaxChannels() const { return max_channels_; }
private:
SctpTransportState state_ = SctpTransportState::kNew;
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
std::optional<double> max_message_size_;
std::optional<int> max_channels_;
};
class SctpTransportObserverInterface {
public:
// This callback carries information about the state of the transport.
// The argument is a pass-by-value snapshot of the state.
// The callback will be called on the network thread.
virtual void OnStateChange(SctpTransportInformation info) = 0;
protected:
virtual ~SctpTransportObserverInterface() = default;
};
// A SCTP transport, as represented to the outside world.
// This object is created on the network thread, and can only be
// accessed on that thread, except for functions explicitly marked otherwise.
// References can be held by other threads, and destruction can therefore
// be initiated by other threads.
class SctpTransportInterface : public webrtc::RefCountInterface {
public:
// This function can be called from other threads.
virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const = 0;
// Returns information on the state of the SctpTransport.
// This function can be called from other threads.
virtual SctpTransportInformation Information() const = 0;
// Observer management.
virtual void RegisterObserver(SctpTransportObserverInterface* observer) = 0;
virtual void UnregisterObserver() = 0;
};
// The size of the SCTP association send buffer. 256kB, the usrsctp default.
constexpr int kSctpSendBufferSize = 256 * 1024;
// SCTP options negotiated in the SDP.
struct SctpOptions {
// https://www.rfc-editor.org/rfc/rfc8841.html#name-sctp-port
// `local_port` and `remote_port` are passed along the wire and the
// listener and connector must be using the same port. They are not related
// to the ports at the IP level. If set to -1 we default to
// kSctpDefaultPort.
// TODO(bugs.webrtc.org/402429107): make these optional<uint16_t>.
int local_port = -1;
int remote_port = -1;
// https://www.rfc-editor.org/rfc/rfc8841.html#name-max-message-size
// `max_message_size` sets the maxium message size on the connection.
// It must be smaller than or equal to kSctpSendBufferSize.
int max_message_size = kSctpSendBufferSize;
};
} // namespace webrtc
#endif // API_SCTP_TRANSPORT_INTERFACE_H_