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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 WEBRTC_P2P_BASE_PORT_H_
#define WEBRTC_P2P_BASE_PORT_H_
#include <map>
#include <set>
#include <string>
#include <vector>
#include "webrtc/p2p/base/candidate.h"
#include "webrtc/p2p/base/packetsocketfactory.h"
#include "webrtc/p2p/base/portinterface.h"
#include "webrtc/p2p/base/stun.h"
#include "webrtc/p2p/base/stunrequest.h"
#include "webrtc/p2p/base/transport.h"
#include "webrtc/base/asyncpacketsocket.h"
#include "webrtc/base/network.h"
#include "webrtc/base/proxyinfo.h"
#include "webrtc/base/ratetracker.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/socketaddress.h"
#include "webrtc/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[];
extern const char UDP_PROTOCOL_NAME[];
extern const char TCP_PROTOCOL_NAME[];
extern const char SSLTCP_PROTOCOL_NAME[];
// 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[];
// The length of time we wait before timing out readability on a connection.
const uint32 CONNECTION_READ_TIMEOUT = 30 * 1000; // 30 seconds
// The length of time we wait before timing out writability on a connection.
const uint32 CONNECTION_WRITE_TIMEOUT = 15 * 1000; // 15 seconds
// The length of time we wait before we become unwritable.
const uint32 CONNECTION_WRITE_CONNECT_TIMEOUT = 5 * 1000; // 5 seconds
// The number of pings that must fail to respond before we become unwritable.
const uint32 CONNECTION_WRITE_CONNECT_FAILURES = 5;
// This is the length of time that we wait for a ping response to come back.
const int CONNECTION_RESPONSE_TIMEOUT = 5 * 1000; // 5 seconds
enum RelayType {
RELAY_GTURN, // Legacy google relay service.
RELAY_TURN // Standard (TURN) relay service.
};
enum IcePriorityValue {
// The reason we are choosing Relay preference 2 is because, we can run
// Relay from client to server on UDP/TCP/TLS. To distinguish the transport
// protocol, we prefer UDP over TCP over TLS.
// For UDP ICE_TYPE_PREFERENCE_RELAY will be 2.
// For TCP ICE_TYPE_PREFERENCE_RELAY will be 1.
// For TLS ICE_TYPE_PREFERENCE_RELAY will be 0.
// Check turnport.cc for setting these values.
ICE_TYPE_PREFERENCE_RELAY = 2,
ICE_TYPE_PREFERENCE_HOST_TCP = 90,
ICE_TYPE_PREFERENCE_SRFLX = 100,
ICE_TYPE_PREFERENCE_PRFLX = 110,
ICE_TYPE_PREFERENCE_HOST = 126
};
const char* ProtoToString(ProtocolType proto);
bool StringToProto(const char* value, ProtocolType* proto);
struct ProtocolAddress {
rtc::SocketAddress address;
ProtocolType proto;
bool secure;
ProtocolAddress(const rtc::SocketAddress& a, ProtocolType p)
: address(a), proto(p), secure(false) { }
ProtocolAddress(const rtc::SocketAddress& a, ProtocolType p, bool sec)
: address(a), proto(p), secure(sec) { }
};
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:
Port(rtc::Thread* thread,
rtc::PacketSocketFactory* factory,
rtc::Network* network,
const rtc::IPAddress& ip,
const std::string& username_fragment,
const std::string& password);
Port(rtc::Thread* thread,
const std::string& type,
rtc::PacketSocketFactory* factory,
rtc::Network* network,
const rtc::IPAddress& ip,
uint16 min_port,
uint16 max_port,
const std::string& username_fragment,
const std::string& password);
virtual ~Port();
virtual const std::string& Type() const { return type_; }
virtual rtc::Network* Network() const { return network_; }
// This method will set the flag which enables standard ICE/STUN procedures
// in STUN connectivity checks. Currently this method does
// 1. Add / Verify MI attribute in STUN binding requests.
// 2. Username attribute in STUN binding request will be RFRAF:LFRAG,
// as opposed to RFRAGLFRAG.
virtual void SetIceProtocolType(IceProtocolType protocol) {
ice_protocol_ = protocol;
}
virtual IceProtocolType IceProtocol() const { return ice_protocol_; }
// Methods to set/get ICE role and tiebreaker values.
IceRole GetIceRole() const { return ice_role_; }
void SetIceRole(IceRole role) { ice_role_ = role; }
void SetIceTiebreaker(uint64 tiebreaker) { tiebreaker_ = tiebreaker; }
uint64 IceTiebreaker() const { return tiebreaker_; }
virtual bool SharedSocket() const { return shared_socket_; }
void ResetSharedSocket() { shared_socket_ = false; }
// 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 generation() { return generation_; }
void set_generation(uint32 generation) { generation_ = generation; }
// ICE requires a single username/password per content/media line. So the
// |ice_username_fragment_| of the ports that belongs to the same content will
// be the same. However this causes a small complication with our relay
// server, which expects different username for RTP and RTCP.
//
// To resolve this problem, we implemented the username_fragment(),
// which returns a different username (calculated from
// |ice_username_fragment_|) for RTCP in the case of ICEPROTO_GOOGLE. And the
// username_fragment() simply returns |ice_username_fragment_| when running
// in ICEPROTO_RFC5245.
//
// As a result the ICEPROTO_GOOGLE will use different usernames for RTP and
// RTCP. And the ICEPROTO_RFC5245 will use same username for both RTP and
// RTCP.
const std::string username_fragment() const;
const std::string& password() const { return 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.
virtual const std::vector<Candidate>& Candidates() const {
return candidates_;
}
// 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.
virtual Connection* GetConnection(
const rtc::SocketAddress& remote_addr);
// 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) {
ASSERT(false);
return false;
}
// 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.
virtual void SendBindingResponse(StunMessage* request,
const rtc::SocketAddress& addr);
virtual void SendBindingErrorResponse(
StunMessage* request, const rtc::SocketAddress& addr,
int error_code, const std::string& reason);
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_; }
virtual void EnablePortPackets();
// Called if the port has no connections and is no longer useful.
void Destroy();
virtual void OnMessage(rtc::Message *pmsg);
// Debugging description of this port
virtual std::string ToString() const;
rtc::IPAddress& ip() { return ip_; }
uint16 min_port() { return min_port_; }
uint16 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,
IceProtocolType* remote_protocol_type) 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 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);
// Returns if RFC 5245 ICE protocol is used.
bool IsStandardIce() const;
// Returns if Google ICE protocol is used.
bool IsGoogleIce() const;
// Returns if Hybrid ICE protocol is used.
bool IsHybridIce() const;
void set_candidate_filter(uint32 candidate_filter) {
candidate_filter_ = candidate_filter;
}
protected:
enum {
MSG_CHECKTIMEOUT = 0,
MSG_FIRST_AVAILABLE
};
void set_type(const std::string& type) { type_ = type; }
void AddAddress(const rtc::SocketAddress& address,
const rtc::SocketAddress& base_address,
const rtc::SocketAddress& related_address,
const std::string& protocol, const std::string& tcptype,
const std::string& type, uint32 type_preference,
uint32 relay_preference, bool final);
// Adds the given connection to the list. (Deleting removes them.)
void AddConnection(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,
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;
}
uint32 candidate_filter() { return candidate_filter_; }
private:
void Construct();
// Called when one of our connections deletes itself.
void OnConnectionDestroyed(Connection* conn);
// Checks if this port is useless, and hence, should be destroyed.
void CheckTimeout();
rtc::Thread* thread_;
rtc::PacketSocketFactory* factory_;
std::string type_;
bool send_retransmit_count_attribute_;
rtc::Network* network_;
rtc::IPAddress ip_;
uint16 min_port_;
uint16 max_port_;
std::string content_name_;
int component_;
uint32 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_;
IceProtocolType ice_protocol_;
IceRole ice_role_;
uint64 tiebreaker_;
bool shared_socket_;
// Information to use when going through a proxy.
std::string user_agent_;
rtc::ProxyInfo proxy_;
// Candidate filter is pushed down to Port such that each Port could
// make its own decision on how to create candidates. For example,
// when IceTransportsType is set to relay, both RelayPort and
// TurnPort will hide raddr to avoid local address leakage.
uint32 candidate_filter_;
friend class Connection;
};
// Represents a communication link between a port on the local client and a
// port on the remote client.
class Connection : public rtc::MessageHandler,
public sigslot::has_slots<> {
public:
// States are from RFC 5245. http://tools.ietf.org/html/rfc5245#section-5.7.4
enum State {
STATE_WAITING = 0, // Check has not been performed, Waiting pair on CL.
STATE_INPROGRESS, // Check has been sent, transaction is in progress.
STATE_SUCCEEDED, // Check already done, produced a successful result.
STATE_FAILED // Check for this connection failed.
};
virtual ~Connection();
// The local port where this connection sends and receives packets.
Port* port() { return port_; }
const Port* port() const { return port_; }
// Returns the description of the local port
virtual const Candidate& local_candidate() const;
// Returns the description of the remote port to which we communicate.
const Candidate& remote_candidate() const { return remote_candidate_; }
// Returns the pair priority.
uint64 priority() const;
enum ReadState {
STATE_READ_INIT = 0, // we have yet to receive a ping
STATE_READABLE = 1, // we have received pings recently
STATE_READ_TIMEOUT = 2, // we haven't received pings in a while
};
ReadState read_state() const { return read_state_; }
bool readable() const { return read_state_ == STATE_READABLE; }
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; }
// Determines whether the connection has finished connecting. This can only
// be false for TCP connections.
bool connected() const { return connected_; }
// Estimate of the round-trip time over this connection.
uint32 rtt() const { return rtt_; }
size_t sent_total_bytes();
size_t sent_bytes_second();
// Used to track how many packets are discarded in the application socket due
// to errors.
size_t sent_discarded_packets();
size_t sent_total_packets();
size_t recv_total_bytes();
size_t recv_bytes_second();
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_remote_ice_mode(IceMode mode) {
remote_ice_mode_ = mode;
}
// Makes the connection go away.
void Destroy();
// 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(uint32 now);
// Called when this connection should try checking writability again.
uint32 last_ping_sent() const { return last_ping_sent_; }
void Ping(uint32 now);
// Called whenever a valid ping is received on this connection. This is
// public because the connection intercepts the first ping for us.
uint32 last_ping_received() const { return last_ping_received_; }
void ReceivedPing();
// Debugging description of this connection
std::string ToDebugId() const;
std::string ToString() const;
std::string ToSensitiveString() const;
bool reported() const { return reported_; }
void set_reported(bool reported) { reported_ = reported;}
// This flag will be set if this connection is the chosen one for media
// transmission. This connection will send STUN ping with USE-CANDIDATE
// attribute.
sigslot::signal1<Connection*> SignalUseCandidate;
// Invoked when Connection receives STUN error response with 487 code.
void HandleRoleConflictFromPeer();
State state() const { return state_; }
IceMode remote_ice_mode() const { return remote_ice_mode_; }
protected:
// 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
void OnConnectionRequestResponse(ConnectionRequest* req,
StunMessage* response);
void OnConnectionRequestErrorResponse(ConnectionRequest* req,
StunMessage* response);
void OnConnectionRequestTimeout(ConnectionRequest* req);
// Changes the state and signals if necessary.
void set_read_state(ReadState value);
void set_write_state(WriteState value);
void set_state(State state);
void set_connected(bool value);
// Checks if this connection is useless, and hence, should be destroyed.
void CheckTimeout();
void OnMessage(rtc::Message *pmsg);
Port* port_;
size_t local_candidate_index_;
Candidate remote_candidate_;
ReadState read_state_;
WriteState write_state_;
bool connected_;
bool pruned_;
// By default |use_candidate_attr_| flag will be true,
// as we will be using agrressive 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_;
IceMode remote_ice_mode_;
StunRequestManager requests_;
uint32 rtt_;
uint32 last_ping_sent_; // last time we sent a ping to the other side
uint32 last_ping_received_; // last time we received a ping from the other
// side
uint32 last_data_received_;
uint32 last_ping_response_received_;
std::vector<uint32> pings_since_last_response_;
rtc::RateTracker recv_rate_tracker_;
rtc::RateTracker send_rate_tracker_;
uint32 sent_packets_discarded_;
uint32 sent_packets_total_;
private:
void MaybeAddPrflxCandidate(ConnectionRequest* request,
StunMessage* response);
bool reported_;
State state_;
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& candidate);
virtual int Send(const void* data, size_t size,
const rtc::PacketOptions& options);
virtual int GetError() { return error_; }
private:
int error_;
};
} // namespace cricket
#endif // WEBRTC_P2P_BASE_PORT_H_