chromium/third_party/webrtc/pc/srtp_transport.h

/*
 *  Copyright 2017 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_SRTP_TRANSPORT_H_
#define PC_SRTP_TRANSPORT_H_

#include <stddef.h>

#include <cstdint>
#include <memory>
#include <string>
#include <vector>

#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/rtc_error.h"
#include "p2p/base/packet_transport_internal.h"
#include "pc/rtp_transport.h"
#include "pc/srtp_session.h"
#include "rtc_base/async_packet_socket.h"
#include "rtc_base/buffer.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network_route.h"

namespace webrtc {

// This subclass of the RtpTransport is used for SRTP which is reponsible for
// protecting/unprotecting the packets. It provides interfaces to set the crypto
// parameters for the SrtpSession underneath.
class SrtpTransport : public RtpTransport { … };

}  // namespace webrtc

#endif  // PC_SRTP_TRANSPORT_H_