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/* Copyright 2018 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_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_
#define API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_
#include <memory>
#include <string>
#include <utility>
#include "absl/types/optional.h"
namespace webrtc {
// Media transport config is made available to both transport and audio / video
// layers, but access to individual interfaces should not be open without
// necessity.
struct MediaTransportConfig {
// Default constructor for no-media transport scenarios.
MediaTransportConfig() = default;
// Constructor for datagram transport scenarios.
explicit MediaTransportConfig(size_t rtp_max_packet_size);
std::string DebugString() const;
// If provided, limits RTP packet size (excludes ICE, IP or network overhead).
absl::optional<size_t> rtp_max_packet_size;
};
} // namespace webrtc
#endif // API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_
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