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/*
* Copyright 2004 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_PORT_H_
#define P2P_BASE_PORT_H_
#include <map>
#include <memory>
#include <set>
#include <string>
#include <vector>
Reland "Clean up libjingle API dependencies." This is a reland of 9185aca9ce1f66e983d9a5e797cab77a64cc46b0 > Original change's description: > > > Clean up libjingle API dependencies. > > > > > > This CL moves candidate.h into the public API, since it has > > > been implicitly included before. > > > > > > This is a straightforward way of solving the circular > > > dependencies involving that file. For instance, > > > libjingle_peerconnection_api includes candidate.h from > > > jsepicecandidate.h, but _api can't depend on rtc_p2p, which > > > depends on _api. In fact, _api can't depend on much at all > > > since it's a very high level abstraction; instead, things > > > should depend on it. > > > > > > Furthermore, we have the case where deprecated headers > > > include headers in internal modules. I just have to turn > > > off include checking for those, but that's not a big deal. > > > > > > This CL punts the problem of callfactoryinterface.h being > > > implicitly included, and pulling in most of the call > > > module with it. This should be addressed in a follow-up > > > CL. > Bug: webrtc:7504 > Change-Id: Icae0ba1a0488550e2871cc65e66d3661707aa5b6 > Reviewed-on: https://webrtc-review.googlesource.com/6460 > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Commit-Queue: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20156} TBR=deadbeef@webrtc.org Bug: webrtc:7504 Change-Id: Ic6598ac2af9355b60bbd289c86dc75e0ae9fed2e Reviewed-on: https://webrtc-review.googlesource.com/6801 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20167}
2017-10-05 14:53:33 +02:00
#include "api/candidate.h"
#include "api/optional.h"
#include "api/rtcerror.h"
#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
#include "logging/rtc_event_log/icelogger.h"
#include "p2p/base/candidatepairinterface.h"
#include "p2p/base/p2pconstants.h"
#include "p2p/base/packetlossestimator.h"
#include "p2p/base/packetsocketfactory.h"
#include "p2p/base/portinterface.h"
#include "p2p/base/stun.h"
#include "p2p/base/stunrequest.h"
#include "rtc_base/asyncpacketsocket.h"
#include "rtc_base/checks.h"
#include "rtc_base/nethelper.h"
#include "rtc_base/network.h"
#include "rtc_base/proxyinfo.h"
#include "rtc_base/ratetracker.h"
#include "rtc_base/sigslot.h"
#include "rtc_base/socketaddress.h"
#include "rtc_base/thread.h"
namespace cricket {
class Connection;
class ConnectionRequest;
extern const char LOCAL_PORT_TYPE[];
extern const char STUN_PORT_TYPE[];
extern const char PRFLX_PORT_TYPE[];
extern const char RELAY_PORT_TYPE[];
// RFC 6544, TCP candidate encoding rules.
extern const int DISCARD_PORT;
extern const char TCPTYPE_ACTIVE_STR[];
extern const char TCPTYPE_PASSIVE_STR[];
extern const char TCPTYPE_SIMOPEN_STR[];
enum RelayType {
RELAY_GTURN, // Legacy google relay service.
RELAY_TURN // Standard (TURN) relay service.
};
enum IcePriorityValue {
ICE_TYPE_PREFERENCE_RELAY_TLS = 0,
ICE_TYPE_PREFERENCE_RELAY_TCP = 1,
ICE_TYPE_PREFERENCE_RELAY_UDP = 2,
ICE_TYPE_PREFERENCE_PRFLX_TCP = 80,
ICE_TYPE_PREFERENCE_HOST_TCP = 90,
ICE_TYPE_PREFERENCE_SRFLX = 100,
ICE_TYPE_PREFERENCE_PRFLX = 110,
ICE_TYPE_PREFERENCE_HOST = 126
};
// 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 port of a STUN candidate.
class StunStats {
public:
StunStats() = default;
StunStats(const StunStats&) = default;
~StunStats() = default;
StunStats& operator=(const StunStats& other) = default;
int stun_binding_requests_sent = 0;
int stun_binding_responses_received = 0;
double stun_binding_rtt_ms_total = 0;
double stun_binding_rtt_ms_squared_total = 0;
};
// Stats that we can return about a candidate.
class CandidateStats {
public:
CandidateStats();
explicit CandidateStats(Candidate candidate);
CandidateStats(const CandidateStats&);
~CandidateStats();
Candidate candidate;
// STUN port stats if this candidate is a STUN candidate.
rtc::Optional<StunStats> stun_stats;
};
typedef std::vector<CandidateStats> CandidateStatsList;
// 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?
bool new_connection; // Is this a newly created connection?
size_t rtt; // The STUN RTT for this connection.
size_t sent_total_bytes; // Total bytes sent on this connection.
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.
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 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
rtc::Optional<uint32_t> current_round_trip_time_ms;
};
// Information about all the candidate pairs of a channel.
typedef std::vector<ConnectionInfo> ConnectionInfos;
const char* ProtoToString(ProtocolType proto);
bool StringToProto(const char* value, ProtocolType* proto);
struct ProtocolAddress {
rtc::SocketAddress address;
ProtocolType proto;
ProtocolAddress(const rtc::SocketAddress& a, ProtocolType p)
: address(a), proto(p) {}
bool operator==(const ProtocolAddress& o) const {
return address == o.address && proto == o.proto;
}
bool operator!=(const ProtocolAddress& o) const { return !(*this == o); }
};
typedef std::set<rtc::SocketAddress> ServerAddresses;
// Represents a local communication mechanism that can be used to create
// connections to similar mechanisms of the other client. Subclasses of this
// one add support for specific mechanisms like local UDP ports.
class Port : public PortInterface, public rtc::MessageHandler,
public sigslot::has_slots<> {
public:
// INIT: The state when a port is just created.
// KEEP_ALIVE_UNTIL_PRUNED: A port should not be destroyed even if no
// connection is using it.
// PRUNED: It will be destroyed if no connection is using it for a period of
// 30 seconds.
enum class State { INIT, KEEP_ALIVE_UNTIL_PRUNED, PRUNED };
Port(rtc::Thread* thread,
const std::string& type,
rtc::PacketSocketFactory* factory,
rtc::Network* network,
const std::string& username_fragment,
const std::string& password);
Make Port (and subclasses) fully "Network"-based, instead of IP-based. For ICE, we want sockets that are bound to specific network interfaces, rather than to specific IP addresses. So, a while ago, we added a "Network" class that gets passed into the Port constructor, in addition to the IP address as before. But we never finished the job of removing the IP address field, such that a Port only guarantees something about the network interface it's associated with, and not the specific IP address it ends up with. This CL does that, and as a consequence, if a port ends up bound to an IP address other than the "best" one (returned by Network::GetBestIP), this *won't* be treated as an error. This is relevant to Android, where even though we pass an IP address into "Bind" as a way of identifying the network, the socket actually gets bound using "android_setsocknetwork", which doesn't provide any guarantees about the IP address. So, if a network interface has multiple IPv6 addresses (for instance), we may not correctly predict the one the OS will choose, and that's ok. This CL also moves "SetAlternateLocalAddress" from VirtualSocket to VirtualSocketServer, which makes for much more readable test code. The next step, if there is one, is to pass along the Network class all the way to SocketServer::Bind. Then the socket server could do smart things with the network information. We could even stick a platform- specific network handle in the Network object, such that the socket server could use it for the binding, or for "sendmsg", for example. See bug 7026 for more context about the sendmsg idea. BUG=webrtc:7715 Review-Url: https://codereview.webrtc.org/2989303002 Cr-Commit-Position: refs/heads/master@{#19251}
2017-08-04 15:01:57 -07:00
// TODO(deadbeef): Delete this constructor once clients are moved off of it.
Port(rtc::Thread* thread,
const std::string& type,
rtc::PacketSocketFactory* factory,
rtc::Network* network,
const rtc::IPAddress& ip,
Make Port (and subclasses) fully "Network"-based, instead of IP-based. For ICE, we want sockets that are bound to specific network interfaces, rather than to specific IP addresses. So, a while ago, we added a "Network" class that gets passed into the Port constructor, in addition to the IP address as before. But we never finished the job of removing the IP address field, such that a Port only guarantees something about the network interface it's associated with, and not the specific IP address it ends up with. This CL does that, and as a consequence, if a port ends up bound to an IP address other than the "best" one (returned by Network::GetBestIP), this *won't* be treated as an error. This is relevant to Android, where even though we pass an IP address into "Bind" as a way of identifying the network, the socket actually gets bound using "android_setsocknetwork", which doesn't provide any guarantees about the IP address. So, if a network interface has multiple IPv6 addresses (for instance), we may not correctly predict the one the OS will choose, and that's ok. This CL also moves "SetAlternateLocalAddress" from VirtualSocket to VirtualSocketServer, which makes for much more readable test code. The next step, if there is one, is to pass along the Network class all the way to SocketServer::Bind. Then the socket server could do smart things with the network information. We could even stick a platform- specific network handle in the Network object, such that the socket server could use it for the binding, or for "sendmsg", for example. See bug 7026 for more context about the sendmsg idea. BUG=webrtc:7715 Review-Url: https://codereview.webrtc.org/2989303002 Cr-Commit-Position: refs/heads/master@{#19251}
2017-08-04 15:01:57 -07:00
const std::string& username_fragment,
const std::string& password);
Make Port (and subclasses) fully "Network"-based, instead of IP-based. For ICE, we want sockets that are bound to specific network interfaces, rather than to specific IP addresses. So, a while ago, we added a "Network" class that gets passed into the Port constructor, in addition to the IP address as before. But we never finished the job of removing the IP address field, such that a Port only guarantees something about the network interface it's associated with, and not the specific IP address it ends up with. This CL does that, and as a consequence, if a port ends up bound to an IP address other than the "best" one (returned by Network::GetBestIP), this *won't* be treated as an error. This is relevant to Android, where even though we pass an IP address into "Bind" as a way of identifying the network, the socket actually gets bound using "android_setsocknetwork", which doesn't provide any guarantees about the IP address. So, if a network interface has multiple IPv6 addresses (for instance), we may not correctly predict the one the OS will choose, and that's ok. This CL also moves "SetAlternateLocalAddress" from VirtualSocket to VirtualSocketServer, which makes for much more readable test code. The next step, if there is one, is to pass along the Network class all the way to SocketServer::Bind. Then the socket server could do smart things with the network information. We could even stick a platform- specific network handle in the Network object, such that the socket server could use it for the binding, or for "sendmsg", for example. See bug 7026 for more context about the sendmsg idea. BUG=webrtc:7715 Review-Url: https://codereview.webrtc.org/2989303002 Cr-Commit-Position: refs/heads/master@{#19251}
2017-08-04 15:01:57 -07:00
Port(rtc::Thread* thread,
const std::string& type,
rtc::PacketSocketFactory* factory,
rtc::Network* network,
uint16_t min_port,
uint16_t max_port,
const std::string& username_fragment,
const std::string& password);
~Port() override;
// Note that the port type does NOT uniquely identify different subclasses of
// Port. Use the 2-tuple of the port type AND the protocol (GetProtocol()) to
// uniquely identify subclasses. Whenever a new subclass of Port introduces a
// conflit in the value of the 2-tuple, make sure that the implementation that
// relies on this 2-tuple for RTTI is properly changed.
const std::string& Type() const override;
rtc::Network* Network() const override;
// Methods to set/get ICE role and tiebreaker values.
IceRole GetIceRole() const override;
void SetIceRole(IceRole role) override;
void SetIceTiebreaker(uint64_t tiebreaker) override;
uint64_t IceTiebreaker() const override;
bool SharedSocket() const override;
void ResetSharedSocket() { shared_socket_ = false; }
// Should not destroy the port even if no connection is using it. Called when
// a port is ready to use.
void KeepAliveUntilPruned();
// Allows a port to be destroyed if no connection is using it.
void Prune();
// The thread on which this port performs its I/O.
rtc::Thread* thread() { return thread_; }
// The factory used to create the sockets of this port.
rtc::PacketSocketFactory* socket_factory() const { return factory_; }
void set_socket_factory(rtc::PacketSocketFactory* factory) {
factory_ = factory;
}
// For debugging purposes.
const std::string& content_name() const { return content_name_; }
void set_content_name(const std::string& content_name) {
content_name_ = content_name;
}
int component() const { return component_; }
void set_component(int component) { component_ = component; }
bool send_retransmit_count_attribute() const {
return send_retransmit_count_attribute_;
}
void set_send_retransmit_count_attribute(bool enable) {
send_retransmit_count_attribute_ = enable;
}
// Identifies the generation that this port was created in.
uint32_t generation() const { return generation_; }
void set_generation(uint32_t generation) { generation_ = generation; }
const std::string username_fragment() const;
const std::string& password() const { return password_; }
// May be called when this port was initially created by a pooled
// PortAllocatorSession, and is now being assigned to an ICE transport.
// Updates the information for candidates as well.
void SetIceParameters(int component,
const std::string& username_fragment,
const std::string& password);
// Fired when candidates are discovered by the port. When all candidates
// are discovered that belong to port SignalAddressReady is fired.
sigslot::signal2<Port*, const Candidate&> SignalCandidateReady;
// Provides all of the above information in one handy object.
const std::vector<Candidate>& Candidates() const override;
// SignalPortComplete is sent when port completes the task of candidates
// allocation.
sigslot::signal1<Port*> SignalPortComplete;
// This signal sent when port fails to allocate candidates and this port
// can't be used in establishing the connections. When port is in shared mode
// and port fails to allocate one of the candidates, port shouldn't send
// this signal as other candidates might be usefull in establishing the
// connection.
sigslot::signal1<Port*> SignalPortError;
// Returns a map containing all of the connections of this port, keyed by the
// remote address.
typedef std::map<rtc::SocketAddress, Connection*> AddressMap;
const AddressMap& connections() { return connections_; }
// Returns the connection to the given address or NULL if none exists.
Connection* GetConnection(const rtc::SocketAddress& remote_addr) override;
// Called each time a connection is created.
sigslot::signal2<Port*, Connection*> SignalConnectionCreated;
// In a shared socket mode each port which shares the socket will decide
// to accept the packet based on the |remote_addr|. Currently only UDP
// port implemented this method.
// TODO(mallinath) - Make it pure virtual.
virtual bool HandleIncomingPacket(rtc::AsyncPacketSocket* socket,
const char* data,
size_t size,
const rtc::SocketAddress& remote_addr,
const rtc::PacketTime& packet_time);
// Shall the port handle packet from this |remote_addr|.
// This method is overridden by TurnPort.
virtual bool CanHandleIncomingPacketsFrom(
const rtc::SocketAddress& remote_addr) const;
// Sends a response message (normal or error) to the given request. One of
// these methods should be called as a response to SignalUnknownAddress.
// NOTE: You MUST call CreateConnection BEFORE SendBindingResponse.
void SendBindingResponse(StunMessage* request,
const rtc::SocketAddress& addr) override;
void SendBindingErrorResponse(StunMessage* request,
const rtc::SocketAddress& addr,
int error_code,
const std::string& reason) override;
void set_proxy(const std::string& user_agent,
const rtc::ProxyInfo& proxy) {
user_agent_ = user_agent;
proxy_ = proxy;
}
const std::string& user_agent() { return user_agent_; }
const rtc::ProxyInfo& proxy() { return proxy_; }
void EnablePortPackets() override;
// Called if the port has no connections and is no longer useful.
void Destroy();
void OnMessage(rtc::Message* pmsg) override;
// Debugging description of this port
std::string ToString() const override;
uint16_t min_port() { return min_port_; }
uint16_t max_port() { return max_port_; }
// Timeout shortening function to speed up unit tests.
void set_timeout_delay(int delay) { timeout_delay_ = delay; }
// This method will return local and remote username fragements from the
// stun username attribute if present.
bool ParseStunUsername(const StunMessage* stun_msg,
std::string* local_username,
std::string* remote_username) const;
void CreateStunUsername(const std::string& remote_username,
std::string* stun_username_attr_str) const;
bool MaybeIceRoleConflict(const rtc::SocketAddress& addr,
IceMessage* stun_msg,
const std::string& remote_ufrag);
// Called when a packet has been sent to the socket.
// This is made pure virtual to notify subclasses of Port that they MUST
// listen to AsyncPacketSocket::SignalSentPacket and then call
// PortInterface::OnSentPacket.
virtual void OnSentPacket(rtc::AsyncPacketSocket* socket,
const rtc::SentPacket& sent_packet) = 0;
// Called when the socket is currently able to send.
void OnReadyToSend();
// Called when the Connection discovers a local peer reflexive candidate.
// Returns the index of the new local candidate.
size_t AddPrflxCandidate(const Candidate& local);
int16_t network_cost() const { return network_cost_; }
void GetStunStats(rtc::Optional<StunStats>* stats) override{};
protected:
enum { MSG_DESTROY_IF_DEAD = 0, MSG_FIRST_AVAILABLE };
virtual void UpdateNetworkCost();
void set_type(const std::string& type) { type_ = type; }
// Deprecated. Use the AddAddress() method below with "url" instead.
// TODO(zhihuang): Remove this after downstream applications stop using it.
void AddAddress(const rtc::SocketAddress& address,
const rtc::SocketAddress& base_address,
const rtc::SocketAddress& related_address,
const std::string& protocol,
const std::string& relay_protocol,
const std::string& tcptype,
const std::string& type,
uint32_t type_preference,
uint32_t relay_preference,
bool final);
void AddAddress(const rtc::SocketAddress& address,
const rtc::SocketAddress& base_address,
const rtc::SocketAddress& related_address,
const std::string& protocol,
const std::string& relay_protocol,
const std::string& tcptype,
const std::string& type,
uint32_t type_preference,
uint32_t relay_preference,
const std::string& url,
bool final);
// Adds the given connection to the map keyed by the remote candidate address.
// If an existing connection has the same address, the existing one will be
// replaced and destroyed.
void AddOrReplaceConnection(Connection* conn);
// Called when a packet is received from an unknown address that is not
// currently a connection. If this is an authenticated STUN binding request,
// then we will signal the client.
void OnReadPacket(const char* data, size_t size,
const rtc::SocketAddress& addr,
ProtocolType proto);
// If the given data comprises a complete and correct STUN message then the
// return value is true, otherwise false. If the message username corresponds
// with this port's username fragment, msg will contain the parsed STUN
// message. Otherwise, the function may send a STUN response internally.
// remote_username contains the remote fragment of the STUN username.
bool GetStunMessage(const char* data,
size_t size,
const rtc::SocketAddress& addr,
std::unique_ptr<IceMessage>* out_msg,
std::string* out_username);
// Checks if the address in addr is compatible with the port's ip.
bool IsCompatibleAddress(const rtc::SocketAddress& addr);
// Returns default DSCP value.
rtc::DiffServCodePoint DefaultDscpValue() const {
// No change from what MediaChannel set.
return rtc::DSCP_NO_CHANGE;
}
// Extra work to be done in subclasses when a connection is destroyed.
virtual void HandleConnectionDestroyed(Connection* conn) {}
private:
void Construct();
// Called when one of our connections deletes itself.
void OnConnectionDestroyed(Connection* conn);
void OnNetworkTypeChanged(const rtc::Network* network);
rtc::Thread* thread_;
rtc::PacketSocketFactory* factory_;
std::string type_;
bool send_retransmit_count_attribute_;
rtc::Network* network_;
uint16_t min_port_;
uint16_t max_port_;
std::string content_name_;
int component_;
uint32_t generation_;
// In order to establish a connection to this Port (so that real data can be
// sent through), the other side must send us a STUN binding request that is
// authenticated with this username_fragment and password.
// PortAllocatorSession will provide these username_fragment and password.
//
// Note: we should always use username_fragment() instead of using
// |ice_username_fragment_| directly. For the details see the comment on
// username_fragment().
std::string ice_username_fragment_;
std::string password_;
std::vector<Candidate> candidates_;
AddressMap connections_;
int timeout_delay_;
bool enable_port_packets_;
IceRole ice_role_;
uint64_t tiebreaker_;
bool shared_socket_;
// Information to use when going through a proxy.
std::string user_agent_;
rtc::ProxyInfo proxy_;
// A virtual cost perceived by the user, usually based on the network type
// (WiFi. vs. Cellular). It takes precedence over the priority when
// comparing two connections.
int16_t network_cost_;
State state_ = State::INIT;
int64_t last_time_all_connections_removed_ = 0;
friend class Connection;
};
// Represents a communication link between a port on the local client and a
// port on the remote client.
class Connection : public CandidatePairInterface,
public rtc::MessageHandler,
public sigslot::has_slots<> {
public:
struct SentPing {
SentPing(const std::string id, int64_t sent_time, uint32_t nomination)
: id(id), sent_time(sent_time), nomination(nomination) {}
std::string id;
int64_t sent_time;
uint32_t nomination;
};
~Connection() override;
// The local port where this connection sends and receives packets.
Port* port() { return port_; }
const Port* port() const { return port_; }
// Implementation of virtual methods in CandidatePairInterface.
// Returns the description of the local port
const Candidate& local_candidate() const override;
// Returns the description of the remote port to which we communicate.
const Candidate& remote_candidate() const override;
// Returns the pair priority.
uint64_t priority() const;
enum WriteState {
STATE_WRITABLE = 0, // we have received ping responses recently
STATE_WRITE_UNRELIABLE = 1, // we have had a few ping failures
STATE_WRITE_INIT = 2, // we have yet to receive a ping response
STATE_WRITE_TIMEOUT = 3, // we have had a large number of ping failures
};
WriteState write_state() const { return write_state_; }
bool writable() const { return write_state_ == STATE_WRITABLE; }
bool receiving() const { return receiving_; }
// Determines whether the connection has finished connecting. This can only
// be false for TCP connections.
bool connected() const { return connected_; }
bool weak() const { return !(writable() && receiving() && connected()); }
bool active() const {
return write_state_ != STATE_WRITE_TIMEOUT;
}
// A connection is dead if it can be safely deleted.
bool dead(int64_t now) const;
// Estimate of the round-trip time over this connection.
int rtt() const { return rtt_; }
int unwritable_timeout() const;
void set_unwritable_timeout(const rtc::Optional<int>& value_ms) {
unwritable_timeout_ = value_ms;
}
int unwritable_min_checks() const;
void set_unwritable_min_checks(const rtc::Optional<int>& value) {
unwritable_min_checks_ = value;
}
// Gets the |ConnectionInfo| stats, where |best_connection| has not been
// populated (default value false).
ConnectionInfo stats();
sigslot::signal1<Connection*> SignalStateChange;
// Sent when the connection has decided that it is no longer of value. It
// will delete itself immediately after this call.
sigslot::signal1<Connection*> SignalDestroyed;
// The connection can send and receive packets asynchronously. This matches
// the interface of AsyncPacketSocket, which may use UDP or TCP under the
// covers.
virtual int Send(const void* data, size_t size,
const rtc::PacketOptions& options) = 0;
// Error if Send() returns < 0
virtual int GetError() = 0;
sigslot::signal4<Connection*, const char*, size_t, const rtc::PacketTime&>
SignalReadPacket;
sigslot::signal1<Connection*> SignalReadyToSend;
// Called when a packet is received on this connection.
void OnReadPacket(const char* data, size_t size,
const rtc::PacketTime& packet_time);
// Called when the socket is currently able to send.
void OnReadyToSend();
// Called when a connection is determined to be no longer useful to us. We
// still keep it around in case the other side wants to use it. But we can
// safely stop pinging on it and we can allow it to time out if the other
// side stops using it as well.
bool pruned() const { return pruned_; }
void Prune();
bool use_candidate_attr() const { return use_candidate_attr_; }
void set_use_candidate_attr(bool enable);
void set_nomination(uint32_t value) { nomination_ = value; }
uint32_t remote_nomination() const { return remote_nomination_; }
// One or several pairs may be nominated based on if Regular or Aggressive
// Nomination is used. https://tools.ietf.org/html/rfc5245#section-8
// |nominated| is defined both for the controlling or controlled agent based
// on if a nomination has been pinged or acknowledged. The controlled agent
// gets its |remote_nomination_| set when pinged by the controlling agent with
// a nomination value. The controlling agent gets its |acked_nomination_| set
// when receiving a response to a nominating ping.
bool nominated() const { return acked_nomination_ || remote_nomination_; }
// Public for unit tests.
void set_remote_nomination(uint32_t remote_nomination) {
remote_nomination_ = remote_nomination;
}
// Public for unit tests.
uint32_t acked_nomination() const { return acked_nomination_; }
void set_remote_ice_mode(IceMode mode) {
remote_ice_mode_ = mode;
}
int receiving_timeout() const;
void set_receiving_timeout(rtc::Optional<int> receiving_timeout_ms) {
receiving_timeout_ = receiving_timeout_ms;
}
// Makes the connection go away.
void Destroy();
// Makes the connection go away, in a failed state.
void FailAndDestroy();
Reland of Do not delete a connection in the turn port with permission error or refresh error. (patchset #1 id:1 of https://codereview.webrtc.org/2090833002/ ) Reason for revert: The Webrtc waterfall indicates that this revert is not necessary. Original issue's description: > Revert of Do not delete a connection in the turn port with permission error or refresh error. (patchset #6 id:260001 of https://codereview.webrtc.org/2068263003/ ) > > Reason for revert: > It broke webrtc builds. > > Original issue's description: > > Do not delete a connection in the turn port with permission error, refresh error, or binding error. > > > > Even if those error happened, the connection may still be able to receive packets for a while. > > If we delete the connections, all packets arriving will be dropped. > > > > BUG=webrtc:6007 > > R=deadbeef@webrtc.org, pthatcher@webrtc.org > > > > Committed: https://crrev.com/3d77deb29c15bfb8f794ef3413837a0ec0f0c131 > > Cr-Commit-Position: refs/heads/master@{#13262} > > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > TBR=pthatcher@webrtc.org,deadbeef@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:6007 > > Committed: https://crrev.com/3159ffae6b1d5cba2ad972bd3d8074ac85f2c7f9 > Cr-Commit-Position: refs/heads/master@{#13265} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6007 Review-Url: https://codereview.webrtc.org/2090073003 Cr-Commit-Position: refs/heads/master@{#13266}
2016-06-22 16:26:29 -07:00
// Prunes the connection and sets its state to STATE_FAILED,
// It will not be used or send pings although it can still receive packets.
void FailAndPrune();
// Checks that the state of this connection is up-to-date. The argument is
// the current time, which is compared against various timeouts.
void UpdateState(int64_t now);
// Called when this connection should try checking writability again.
int64_t last_ping_sent() const { return last_ping_sent_; }
void Ping(int64_t now);
void ReceivedPingResponse(int rtt, const std::string& request_id);
int64_t last_ping_response_received() const {
return last_ping_response_received_;
}
// Used to check if any STUN ping response has been received.
int rtt_samples() const { return rtt_samples_; }
// Called whenever a valid ping is received on this connection. This is
// public because the connection intercepts the first ping for us.
int64_t last_ping_received() const { return last_ping_received_; }
void ReceivedPing();
// Handles the binding request; sends a response if this is a valid request.
void HandleBindingRequest(IceMessage* msg);
int64_t last_data_received() const { return last_data_received_; }
// Debugging description of this connection
std::string ToDebugId() const;
std::string ToString() const;
std::string ToSensitiveString() const;
// Structured description of this candidate pair.
const webrtc::IceCandidatePairDescription& ToLogDescription();
// Integer typed hash value of this candidate pair.
uint32_t hash() { return hash_; }
void set_ice_event_log(webrtc::IceEventLog* ice_event_log) {
ice_event_log_ = ice_event_log;
}
// Prints pings_since_last_response_ into a string.
void PrintPingsSinceLastResponse(std::string* pings, size_t max);
bool reported() const { return reported_; }
void set_reported(bool reported) { reported_ = reported;}
// This signal will be fired if this connection is nominated by the
// controlling side.
sigslot::signal1<Connection*> SignalNominated;
// Invoked when Connection receives STUN error response with 487 code.
void HandleRoleConflictFromPeer();
IceCandidatePairState state() const { return state_; }
int num_pings_sent() const { return num_pings_sent_; }
IceMode remote_ice_mode() const { return remote_ice_mode_; }
uint32_t ComputeNetworkCost() const;
// Update the ICE password and/or generation of the remote candidate if the
// ufrag in |params| matches the candidate's ufrag, and the
// candidate's password and/or ufrag has not been set.
void MaybeSetRemoteIceParametersAndGeneration(const IceParameters& params,
int generation);
// If |remote_candidate_| is peer reflexive and is equivalent to
// |new_candidate| except the type, update |remote_candidate_| to
// |new_candidate|.
void MaybeUpdatePeerReflexiveCandidate(const Candidate& new_candidate);
// Returns the last received time of any data, stun request, or stun
// response in milliseconds
int64_t last_received() const;
// Returns the last time when the connection changed its receiving state.
int64_t receiving_unchanged_since() const {
return receiving_unchanged_since_;
}
bool stable(int64_t now) const;
protected:
enum { MSG_DELETE = 0, MSG_FIRST_AVAILABLE };
// Constructs a new connection to the given remote port.
Connection(Port* port, size_t index, const Candidate& candidate);
// Called back when StunRequestManager has a stun packet to send
void OnSendStunPacket(const void* data, size_t size, StunRequest* req);
// Callbacks from ConnectionRequest
virtual void OnConnectionRequestResponse(ConnectionRequest* req,
StunMessage* response);
void OnConnectionRequestErrorResponse(ConnectionRequest* req,
StunMessage* response);
void OnConnectionRequestTimeout(ConnectionRequest* req);
void OnConnectionRequestSent(ConnectionRequest* req);
bool rtt_converged() const;
// If the response is not received within 2 * RTT, the response is assumed to
// be missing.
bool missing_responses(int64_t now) const;
// Changes the state and signals if necessary.
void set_write_state(WriteState value);
void UpdateReceiving(int64_t now);
void set_state(IceCandidatePairState state);
void set_connected(bool value);
uint32_t nomination() const { return nomination_; }
void OnMessage(rtc::Message* pmsg) override;
Port* port_;
size_t local_candidate_index_;
Candidate remote_candidate_;
ConnectionInfo stats_;
rtc::RateTracker recv_rate_tracker_;
rtc::RateTracker send_rate_tracker_;
private:
// Update the local candidate based on the mapped address attribute.
// If the local candidate changed, fires SignalStateChange.
void MaybeUpdateLocalCandidate(ConnectionRequest* request,
StunMessage* response);
void LogCandidatePairEvent(webrtc::IceCandidatePairEventType type);
WriteState write_state_;
bool receiving_;
bool connected_;
bool pruned_;
// By default |use_candidate_attr_| flag will be true,
// as we will be using aggressive nomination.
// But when peer is ice-lite, this flag "must" be initialized to false and
// turn on when connection becomes "best connection".
bool use_candidate_attr_;
// Used by the controlling side to indicate that this connection will be
// selected for transmission if the peer supports ICE-renomination when this
// value is positive. A larger-value indicates that a connection is nominated
// later and should be selected by the controlled side with higher precedence.
// A zero-value indicates not nominating this connection.
uint32_t nomination_ = 0;
// The last nomination that has been acknowledged.
uint32_t acked_nomination_ = 0;
// Used by the controlled side to remember the nomination value received from
// the controlling side. When the peer does not support ICE re-nomination,
// its value will be 1 if the connection has been nominated.
uint32_t remote_nomination_ = 0;
IceMode remote_ice_mode_;
StunRequestManager requests_;
int rtt_;
int rtt_samples_ = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-totalroundtriptime
uint64_t total_round_trip_time_ms_ = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-currentroundtriptime
rtc::Optional<uint32_t> current_round_trip_time_ms_;
int64_t last_ping_sent_; // last time we sent a ping to the other side
int64_t last_ping_received_; // last time we received a ping from the other
// side
int64_t last_data_received_;
int64_t last_ping_response_received_;
int64_t receiving_unchanged_since_ = 0;
std::vector<SentPing> pings_since_last_response_;
PacketLossEstimator packet_loss_estimator_;
rtc::Optional<int> unwritable_timeout_;
rtc::Optional<int> unwritable_min_checks_;
bool reported_;
IceCandidatePairState state_;
// Time duration to switch from receiving to not receiving.
rtc::Optional<int> receiving_timeout_;
int64_t time_created_ms_;
int num_pings_sent_ = 0;
rtc::Optional<webrtc::IceCandidatePairDescription> log_description_;
uint32_t hash_;
webrtc::IceEventLog* ice_event_log_ = nullptr;
friend class Port;
friend class ConnectionRequest;
};
// ProxyConnection defers all the interesting work to the port.
class ProxyConnection : public Connection {
public:
ProxyConnection(Port* port, size_t index, const Candidate& remote_candidate);
int Send(const void* data,
size_t size,
const rtc::PacketOptions& options) override;
int GetError() override;
private:
int error_ = 0;
};
} // namespace cricket
#endif // P2P_BASE_PORT_H_