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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 P2P_BASE_CONNECTION_INFO_H_
#define P2P_BASE_CONNECTION_INFO_H_
#include <cstddef>
#include <cstdint>
#include <optional>
#include <vector>
#include "api/candidate.h"
#include "api/units/timestamp.h"
namespace webrtc {
// States are from RFC 5245. http://tools.ietf.org/html/rfc5245#section-5.7.4
enum class IceCandidatePairState {
WAITING = 0, // Check has not been performed, Waiting pair on CL.
IN_PROGRESS, // Check has been sent, transaction is in progress.
SUCCEEDED, // Check already done, produced a successful result.
FAILED, // Check for this connection failed.
// According to spec there should also be a frozen state, but nothing is ever
// frozen because we have not implemented ICE freezing logic.
};
// Stats that we can return about the connections for a transport channel.
// TODO(hta): Rename to ConnectionStats
struct ConnectionInfo {
ConnectionInfo();
ConnectionInfo(const ConnectionInfo&);
~ConnectionInfo();
bool best_connection; // Is this the best connection we have?
bool writable; // Has this connection received a STUN response?
bool receiving; // Has this connection received anything?
bool timeout; // Has this connection timed out?
size_t rtt; // The STUN RTT for this connection.
size_t sent_discarded_bytes; // Number of outgoing bytes discarded due to
// socket errors.
size_t sent_total_bytes; // Total bytes sent on this connection. Does not
// include discarded bytes.
size_t sent_bytes_second; // Bps over the last measurement interval.
size_t sent_discarded_packets; // Number of outgoing packets discarded due to
// socket errors.
size_t sent_total_packets; // Number of total outgoing packets attempted for
// sending, including discarded packets.
size_t sent_ping_requests_total; // Number of STUN ping request sent.
size_t sent_ping_requests_before_first_response; // Number of STUN ping
// sent before receiving the first response.
size_t sent_ping_responses; // Number of STUN ping response sent.
size_t recv_total_bytes; // Total bytes received on this connection.
size_t recv_bytes_second; // Bps over the last measurement interval.
size_t packets_received; // Number of packets that were received.
size_t recv_ping_requests; // Number of STUN ping request received.
size_t recv_ping_responses; // Number of STUN ping response received.
Candidate local_candidate; // The local candidate for this connection.
Candidate remote_candidate; // The remote candidate for this connection.
void* key; // A static value that identifies this conn.
// https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-state
IceCandidatePairState state;
// https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-priority
uint64_t priority;
// https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-nominated
bool nominated;
// https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-totalroundtriptime
uint64_t total_round_trip_time_ms;
// https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-currentroundtriptime
std::optional<uint32_t> current_round_trip_time_ms;
// https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-lastpacketreceivedtimestamp
std::optional<Timestamp> last_data_received;
std::optional<Timestamp> last_data_sent;
};
// Information about all the candidate pairs of a channel.
typedef std::vector<ConnectionInfo> ConnectionInfos;
} // namespace webrtc
// Re-export symbols from the webrtc namespace for backwards compatibility.
// TODO(bugs.webrtc.org/4222596): Remove once all references are updated.
#ifdef WEBRTC_ALLOW_DEPRECATED_NAMESPACES
namespace cricket {
using ::webrtc::ConnectionInfo;
using ::webrtc::ConnectionInfos;
using ::webrtc::IceCandidatePairState;
} // namespace cricket
#endif // WEBRTC_ALLOW_DEPRECATED_NAMESPACES
#endif // P2P_BASE_CONNECTION_INFO_H_
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