blob: 2961355e3e1cf839689eeb72ad143c7f3c7960cb [file] [log] [blame]
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001/*
2 * Copyright 2004 The WebRTC Project Authors. All rights reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11#ifndef WEBRTC_P2P_BASE_PORT_H_
12#define WEBRTC_P2P_BASE_PORT_H_
13
14#include <map>
15#include <set>
16#include <string>
17#include <vector>
18
19#include "webrtc/p2p/base/candidate.h"
Honghai Zhangcc411c02016-03-29 17:27:21 -070020#include "webrtc/p2p/base/candidatepairinterface.h"
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000021#include "webrtc/p2p/base/packetsocketfactory.h"
22#include "webrtc/p2p/base/portinterface.h"
23#include "webrtc/p2p/base/stun.h"
24#include "webrtc/p2p/base/stunrequest.h"
25#include "webrtc/p2p/base/transport.h"
26#include "webrtc/base/asyncpacketsocket.h"
27#include "webrtc/base/network.h"
28#include "webrtc/base/proxyinfo.h"
29#include "webrtc/base/ratetracker.h"
30#include "webrtc/base/sigslot.h"
31#include "webrtc/base/socketaddress.h"
32#include "webrtc/base/thread.h"
33
34namespace cricket {
35
36class Connection;
37class ConnectionRequest;
38
39extern const char LOCAL_PORT_TYPE[];
40extern const char STUN_PORT_TYPE[];
41extern const char PRFLX_PORT_TYPE[];
42extern const char RELAY_PORT_TYPE[];
43
44extern const char UDP_PROTOCOL_NAME[];
45extern const char TCP_PROTOCOL_NAME[];
46extern const char SSLTCP_PROTOCOL_NAME[];
47
48// RFC 6544, TCP candidate encoding rules.
49extern const int DISCARD_PORT;
50extern const char TCPTYPE_ACTIVE_STR[];
51extern const char TCPTYPE_PASSIVE_STR[];
52extern const char TCPTYPE_SIMOPEN_STR[];
53
Honghai Zhang2b342bf2015-09-30 09:51:58 -070054// The minimum time we will wait before destroying a connection after creating
55// it.
honghaiz34b11eb2016-03-16 08:55:44 -070056static const int MIN_CONNECTION_LIFETIME = 10 * 1000; // 10 seconds.
Peter Thatcher04ac81f2015-09-21 11:48:28 -070057
Honghai Zhang2cd7afe2015-11-12 11:14:33 -080058// A connection will be declared dead if it has not received anything for this
59// long.
honghaiz34b11eb2016-03-16 08:55:44 -070060static const int DEAD_CONNECTION_RECEIVE_TIMEOUT = 30 * 1000; // 30 seconds.
Honghai Zhang2cd7afe2015-11-12 11:14:33 -080061
Peter Thatcher04ac81f2015-09-21 11:48:28 -070062// The timeout duration when a connection does not receive anything.
honghaiz34b11eb2016-03-16 08:55:44 -070063static const int WEAK_CONNECTION_RECEIVE_TIMEOUT = 2500; // 2.5 seconds
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000064
65// The length of time we wait before timing out writability on a connection.
honghaiz34b11eb2016-03-16 08:55:44 -070066static const int CONNECTION_WRITE_TIMEOUT = 15 * 1000; // 15 seconds
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000067
68// The length of time we wait before we become unwritable.
honghaiz34b11eb2016-03-16 08:55:44 -070069static const int CONNECTION_WRITE_CONNECT_TIMEOUT = 5 * 1000; // 5 seconds
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000070
71// This is the length of time that we wait for a ping response to come back.
honghaiz34b11eb2016-03-16 08:55:44 -070072static const int CONNECTION_RESPONSE_TIMEOUT = 5 * 1000; // 5 seconds
73
74// The number of pings that must fail to respond before we become unwritable.
75static const uint32_t CONNECTION_WRITE_CONNECT_FAILURES = 5;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000076
77enum RelayType {
78 RELAY_GTURN, // Legacy google relay service.
79 RELAY_TURN // Standard (TURN) relay service.
80};
81
82enum IcePriorityValue {
83 // The reason we are choosing Relay preference 2 is because, we can run
84 // Relay from client to server on UDP/TCP/TLS. To distinguish the transport
85 // protocol, we prefer UDP over TCP over TLS.
86 // For UDP ICE_TYPE_PREFERENCE_RELAY will be 2.
87 // For TCP ICE_TYPE_PREFERENCE_RELAY will be 1.
88 // For TLS ICE_TYPE_PREFERENCE_RELAY will be 0.
89 // Check turnport.cc for setting these values.
90 ICE_TYPE_PREFERENCE_RELAY = 2,
91 ICE_TYPE_PREFERENCE_HOST_TCP = 90,
92 ICE_TYPE_PREFERENCE_SRFLX = 100,
93 ICE_TYPE_PREFERENCE_PRFLX = 110,
94 ICE_TYPE_PREFERENCE_HOST = 126
95};
96
97const char* ProtoToString(ProtocolType proto);
98bool StringToProto(const char* value, ProtocolType* proto);
99
100struct ProtocolAddress {
101 rtc::SocketAddress address;
102 ProtocolType proto;
103 bool secure;
104
105 ProtocolAddress(const rtc::SocketAddress& a, ProtocolType p)
106 : address(a), proto(p), secure(false) { }
107 ProtocolAddress(const rtc::SocketAddress& a, ProtocolType p, bool sec)
108 : address(a), proto(p), secure(sec) { }
109};
110
111typedef std::set<rtc::SocketAddress> ServerAddresses;
112
113// Represents a local communication mechanism that can be used to create
114// connections to similar mechanisms of the other client. Subclasses of this
115// one add support for specific mechanisms like local UDP ports.
116class Port : public PortInterface, public rtc::MessageHandler,
117 public sigslot::has_slots<> {
118 public:
pkasting@chromium.org332331f2014-11-06 20:19:22 +0000119 Port(rtc::Thread* thread,
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000120 rtc::PacketSocketFactory* factory,
pkasting@chromium.org332331f2014-11-06 20:19:22 +0000121 rtc::Network* network,
122 const rtc::IPAddress& ip,
123 const std::string& username_fragment,
124 const std::string& password);
125 Port(rtc::Thread* thread,
126 const std::string& type,
127 rtc::PacketSocketFactory* factory,
128 rtc::Network* network,
129 const rtc::IPAddress& ip,
Peter Boström0c4e06b2015-10-07 12:23:21 +0200130 uint16_t min_port,
131 uint16_t max_port,
pkasting@chromium.org332331f2014-11-06 20:19:22 +0000132 const std::string& username_fragment,
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000133 const std::string& password);
134 virtual ~Port();
135
136 virtual const std::string& Type() const { return type_; }
137 virtual rtc::Network* Network() const { return network_; }
138
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000139 // Methods to set/get ICE role and tiebreaker values.
140 IceRole GetIceRole() const { return ice_role_; }
141 void SetIceRole(IceRole role) { ice_role_ = role; }
142
Peter Boström0c4e06b2015-10-07 12:23:21 +0200143 void SetIceTiebreaker(uint64_t tiebreaker) { tiebreaker_ = tiebreaker; }
144 uint64_t IceTiebreaker() const { return tiebreaker_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000145
146 virtual bool SharedSocket() const { return shared_socket_; }
147 void ResetSharedSocket() { shared_socket_ = false; }
148
149 // The thread on which this port performs its I/O.
150 rtc::Thread* thread() { return thread_; }
151
152 // The factory used to create the sockets of this port.
153 rtc::PacketSocketFactory* socket_factory() const { return factory_; }
154 void set_socket_factory(rtc::PacketSocketFactory* factory) {
155 factory_ = factory;
156 }
157
158 // For debugging purposes.
159 const std::string& content_name() const { return content_name_; }
160 void set_content_name(const std::string& content_name) {
161 content_name_ = content_name;
162 }
163
164 int component() const { return component_; }
165 void set_component(int component) { component_ = component; }
166
167 bool send_retransmit_count_attribute() const {
168 return send_retransmit_count_attribute_;
169 }
170 void set_send_retransmit_count_attribute(bool enable) {
171 send_retransmit_count_attribute_ = enable;
172 }
173
174 // Identifies the generation that this port was created in.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200175 uint32_t generation() { return generation_; }
176 void set_generation(uint32_t generation) { generation_ = generation; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000177
178 // ICE requires a single username/password per content/media line. So the
179 // |ice_username_fragment_| of the ports that belongs to the same content will
180 // be the same. However this causes a small complication with our relay
181 // server, which expects different username for RTP and RTCP.
182 //
183 // To resolve this problem, we implemented the username_fragment(),
184 // which returns a different username (calculated from
185 // |ice_username_fragment_|) for RTCP in the case of ICEPROTO_GOOGLE. And the
186 // username_fragment() simply returns |ice_username_fragment_| when running
187 // in ICEPROTO_RFC5245.
188 //
189 // As a result the ICEPROTO_GOOGLE will use different usernames for RTP and
190 // RTCP. And the ICEPROTO_RFC5245 will use same username for both RTP and
191 // RTCP.
192 const std::string username_fragment() const;
193 const std::string& password() const { return password_; }
194
195 // Fired when candidates are discovered by the port. When all candidates
196 // are discovered that belong to port SignalAddressReady is fired.
197 sigslot::signal2<Port*, const Candidate&> SignalCandidateReady;
198
199 // Provides all of the above information in one handy object.
200 virtual const std::vector<Candidate>& Candidates() const {
201 return candidates_;
202 }
203
204 // SignalPortComplete is sent when port completes the task of candidates
205 // allocation.
206 sigslot::signal1<Port*> SignalPortComplete;
207 // This signal sent when port fails to allocate candidates and this port
208 // can't be used in establishing the connections. When port is in shared mode
209 // and port fails to allocate one of the candidates, port shouldn't send
210 // this signal as other candidates might be usefull in establishing the
211 // connection.
212 sigslot::signal1<Port*> SignalPortError;
213
214 // Returns a map containing all of the connections of this port, keyed by the
215 // remote address.
216 typedef std::map<rtc::SocketAddress, Connection*> AddressMap;
217 const AddressMap& connections() { return connections_; }
218
219 // Returns the connection to the given address or NULL if none exists.
220 virtual Connection* GetConnection(
221 const rtc::SocketAddress& remote_addr);
222
223 // Called each time a connection is created.
224 sigslot::signal2<Port*, Connection*> SignalConnectionCreated;
225
226 // In a shared socket mode each port which shares the socket will decide
227 // to accept the packet based on the |remote_addr|. Currently only UDP
228 // port implemented this method.
229 // TODO(mallinath) - Make it pure virtual.
230 virtual bool HandleIncomingPacket(
231 rtc::AsyncPacketSocket* socket, const char* data, size_t size,
232 const rtc::SocketAddress& remote_addr,
233 const rtc::PacketTime& packet_time) {
234 ASSERT(false);
235 return false;
236 }
237
238 // Sends a response message (normal or error) to the given request. One of
239 // these methods should be called as a response to SignalUnknownAddress.
240 // NOTE: You MUST call CreateConnection BEFORE SendBindingResponse.
241 virtual void SendBindingResponse(StunMessage* request,
242 const rtc::SocketAddress& addr);
243 virtual void SendBindingErrorResponse(
244 StunMessage* request, const rtc::SocketAddress& addr,
245 int error_code, const std::string& reason);
246
247 void set_proxy(const std::string& user_agent,
248 const rtc::ProxyInfo& proxy) {
249 user_agent_ = user_agent;
250 proxy_ = proxy;
251 }
252 const std::string& user_agent() { return user_agent_; }
253 const rtc::ProxyInfo& proxy() { return proxy_; }
254
255 virtual void EnablePortPackets();
256
257 // Called if the port has no connections and is no longer useful.
258 void Destroy();
259
260 virtual void OnMessage(rtc::Message *pmsg);
261
262 // Debugging description of this port
263 virtual std::string ToString() const;
pthatcher@webrtc.org0ba15332015-01-10 00:47:02 +0000264 const rtc::IPAddress& ip() const { return ip_; }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200265 uint16_t min_port() { return min_port_; }
266 uint16_t max_port() { return max_port_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000267
268 // Timeout shortening function to speed up unit tests.
269 void set_timeout_delay(int delay) { timeout_delay_ = delay; }
270
271 // This method will return local and remote username fragements from the
272 // stun username attribute if present.
273 bool ParseStunUsername(const StunMessage* stun_msg,
274 std::string* local_username,
Peter Thatcher7cbd1882015-09-17 18:54:52 -0700275 std::string* remote_username) const;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000276 void CreateStunUsername(const std::string& remote_username,
277 std::string* stun_username_attr_str) const;
278
279 bool MaybeIceRoleConflict(const rtc::SocketAddress& addr,
280 IceMessage* stun_msg,
281 const std::string& remote_ufrag);
282
stefanc1aeaf02015-10-15 07:26:07 -0700283 // Called when a packet has been sent to the socket.
Stefan Holmer55674ff2016-01-14 15:49:16 +0100284 // This is made pure virtual to notify subclasses of Port that they MUST
285 // listen to AsyncPacketSocket::SignalSentPacket and then call
286 // PortInterface::OnSentPacket.
287 virtual void OnSentPacket(rtc::AsyncPacketSocket* socket,
288 const rtc::SentPacket& sent_packet) = 0;
stefanc1aeaf02015-10-15 07:26:07 -0700289
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000290 // Called when the socket is currently able to send.
291 void OnReadyToSend();
292
293 // Called when the Connection discovers a local peer reflexive candidate.
294 // Returns the index of the new local candidate.
295 size_t AddPrflxCandidate(const Candidate& local);
296
Peter Boström0c4e06b2015-10-07 12:23:21 +0200297 void set_candidate_filter(uint32_t candidate_filter) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000298 candidate_filter_ = candidate_filter;
299 }
honghaiza0c44ea2016-03-23 16:07:48 -0700300 int16_t network_cost() const { return network_cost_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000301
302 protected:
303 enum {
honghaizd0b31432015-09-30 12:42:17 -0700304 MSG_DEAD = 0,
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000305 MSG_FIRST_AVAILABLE
306 };
307
308 void set_type(const std::string& type) { type_ = type; }
309
310 void AddAddress(const rtc::SocketAddress& address,
311 const rtc::SocketAddress& base_address,
312 const rtc::SocketAddress& related_address,
Guo-wei Shieh3d564c12015-08-19 16:51:15 -0700313 const std::string& protocol,
314 const std::string& relay_protocol,
315 const std::string& tcptype,
316 const std::string& type,
Peter Boström0c4e06b2015-10-07 12:23:21 +0200317 uint32_t type_preference,
318 uint32_t relay_preference,
Guo-wei Shieh3d564c12015-08-19 16:51:15 -0700319 bool final);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000320
321 // Adds the given connection to the list. (Deleting removes them.)
322 void AddConnection(Connection* conn);
323
324 // Called when a packet is received from an unknown address that is not
325 // currently a connection. If this is an authenticated STUN binding request,
326 // then we will signal the client.
327 void OnReadPacket(const char* data, size_t size,
328 const rtc::SocketAddress& addr,
329 ProtocolType proto);
330
331 // If the given data comprises a complete and correct STUN message then the
332 // return value is true, otherwise false. If the message username corresponds
333 // with this port's username fragment, msg will contain the parsed STUN
334 // message. Otherwise, the function may send a STUN response internally.
335 // remote_username contains the remote fragment of the STUN username.
kwiberg6baec032016-03-15 11:09:39 -0700336 bool GetStunMessage(const char* data,
337 size_t size,
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000338 const rtc::SocketAddress& addr,
kwiberg6baec032016-03-15 11:09:39 -0700339 rtc::scoped_ptr<IceMessage>* out_msg,
340 std::string* out_username);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000341
342 // Checks if the address in addr is compatible with the port's ip.
343 bool IsCompatibleAddress(const rtc::SocketAddress& addr);
344
345 // Returns default DSCP value.
346 rtc::DiffServCodePoint DefaultDscpValue() const {
347 // No change from what MediaChannel set.
348 return rtc::DSCP_NO_CHANGE;
349 }
350
Peter Boström0c4e06b2015-10-07 12:23:21 +0200351 uint32_t candidate_filter() { return candidate_filter_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000352
353 private:
354 void Construct();
355 // Called when one of our connections deletes itself.
356 void OnConnectionDestroyed(Connection* conn);
357
honghaizd0b31432015-09-30 12:42:17 -0700358 // Whether this port is dead, and hence, should be destroyed on the controlled
359 // side.
360 bool dead() const {
361 return ice_role_ == ICEROLE_CONTROLLED && connections_.empty();
362 }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000363
honghaize3c6c822016-02-17 13:00:28 -0800364 void OnNetworkInactive(const rtc::Network* network);
365
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000366 rtc::Thread* thread_;
367 rtc::PacketSocketFactory* factory_;
368 std::string type_;
369 bool send_retransmit_count_attribute_;
370 rtc::Network* network_;
371 rtc::IPAddress ip_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200372 uint16_t min_port_;
373 uint16_t max_port_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000374 std::string content_name_;
375 int component_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200376 uint32_t generation_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000377 // In order to establish a connection to this Port (so that real data can be
378 // sent through), the other side must send us a STUN binding request that is
379 // authenticated with this username_fragment and password.
380 // PortAllocatorSession will provide these username_fragment and password.
381 //
382 // Note: we should always use username_fragment() instead of using
383 // |ice_username_fragment_| directly. For the details see the comment on
384 // username_fragment().
385 std::string ice_username_fragment_;
386 std::string password_;
387 std::vector<Candidate> candidates_;
388 AddressMap connections_;
389 int timeout_delay_;
390 bool enable_port_packets_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000391 IceRole ice_role_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200392 uint64_t tiebreaker_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000393 bool shared_socket_;
394 // Information to use when going through a proxy.
395 std::string user_agent_;
396 rtc::ProxyInfo proxy_;
397
398 // Candidate filter is pushed down to Port such that each Port could
399 // make its own decision on how to create candidates. For example,
400 // when IceTransportsType is set to relay, both RelayPort and
401 // TurnPort will hide raddr to avoid local address leakage.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200402 uint32_t candidate_filter_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000403
honghaize1a0c942016-02-16 14:54:56 -0800404 // A virtual cost perceived by the user, usually based on the network type
405 // (WiFi. vs. Cellular). It takes precedence over the priority when
406 // comparing two connections.
honghaiza0c44ea2016-03-23 16:07:48 -0700407 uint16_t network_cost_;
honghaize1a0c942016-02-16 14:54:56 -0800408
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000409 friend class Connection;
410};
411
412// Represents a communication link between a port on the local client and a
413// port on the remote client.
Honghai Zhangcc411c02016-03-29 17:27:21 -0700414class Connection : public CandidatePairInterface,
415 public rtc::MessageHandler,
416 public sigslot::has_slots<> {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000417 public:
Peter Thatcher1cf6f812015-05-15 10:40:45 -0700418 struct SentPing {
honghaiz34b11eb2016-03-16 08:55:44 -0700419 SentPing(const std::string id, int64_t sent_time)
Peter Boström0c4e06b2015-10-07 12:23:21 +0200420 : id(id), sent_time(sent_time) {}
Peter Thatcher1cf6f812015-05-15 10:40:45 -0700421
422 std::string id;
honghaiz34b11eb2016-03-16 08:55:44 -0700423 int64_t sent_time;
Peter Thatcher1cf6f812015-05-15 10:40:45 -0700424 };
425
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000426 // States are from RFC 5245. http://tools.ietf.org/html/rfc5245#section-5.7.4
427 enum State {
428 STATE_WAITING = 0, // Check has not been performed, Waiting pair on CL.
429 STATE_INPROGRESS, // Check has been sent, transaction is in progress.
430 STATE_SUCCEEDED, // Check already done, produced a successful result.
431 STATE_FAILED // Check for this connection failed.
432 };
433
434 virtual ~Connection();
435
436 // The local port where this connection sends and receives packets.
437 Port* port() { return port_; }
438 const Port* port() const { return port_; }
439
Honghai Zhangcc411c02016-03-29 17:27:21 -0700440 // Implementation of virtual methods in CandidatePairInterface.
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000441 // Returns the description of the local port
442 virtual const Candidate& local_candidate() const;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000443 // Returns the description of the remote port to which we communicate.
Honghai Zhangcc411c02016-03-29 17:27:21 -0700444 virtual const Candidate& remote_candidate() const;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000445
446 // Returns the pair priority.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200447 uint64_t priority() const;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000448
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000449 enum WriteState {
450 STATE_WRITABLE = 0, // we have received ping responses recently
451 STATE_WRITE_UNRELIABLE = 1, // we have had a few ping failures
452 STATE_WRITE_INIT = 2, // we have yet to receive a ping response
453 STATE_WRITE_TIMEOUT = 3, // we have had a large number of ping failures
454 };
455
456 WriteState write_state() const { return write_state_; }
457 bool writable() const { return write_state_ == STATE_WRITABLE; }
Peter Thatcher04ac81f2015-09-21 11:48:28 -0700458 bool receiving() const { return receiving_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000459
460 // Determines whether the connection has finished connecting. This can only
461 // be false for TCP connections.
462 bool connected() const { return connected_; }
Honghai Zhang2b342bf2015-09-30 09:51:58 -0700463 bool weak() const { return !(writable() && receiving() && connected()); }
464 bool active() const {
Honghai Zhang2b342bf2015-09-30 09:51:58 -0700465 return write_state_ != STATE_WRITE_TIMEOUT;
466 }
467 // A connection is dead if it can be safely deleted.
honghaiz34b11eb2016-03-16 08:55:44 -0700468 bool dead(int64_t now) const;
honghaiz89374372015-09-24 13:14:47 -0700469
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000470 // Estimate of the round-trip time over this connection.
honghaiz34b11eb2016-03-16 08:55:44 -0700471 int rtt() const { return rtt_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000472
473 size_t sent_total_bytes();
474 size_t sent_bytes_second();
guoweis@webrtc.org930e0042014-11-17 19:42:14 +0000475 // Used to track how many packets are discarded in the application socket due
476 // to errors.
477 size_t sent_discarded_packets();
478 size_t sent_total_packets();
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000479 size_t recv_total_bytes();
480 size_t recv_bytes_second();
481 sigslot::signal1<Connection*> SignalStateChange;
482
483 // Sent when the connection has decided that it is no longer of value. It
484 // will delete itself immediately after this call.
485 sigslot::signal1<Connection*> SignalDestroyed;
486
487 // The connection can send and receive packets asynchronously. This matches
488 // the interface of AsyncPacketSocket, which may use UDP or TCP under the
489 // covers.
490 virtual int Send(const void* data, size_t size,
491 const rtc::PacketOptions& options) = 0;
492
493 // Error if Send() returns < 0
494 virtual int GetError() = 0;
495
Peter Thatcher04ac81f2015-09-21 11:48:28 -0700496 sigslot::signal4<Connection*, const char*, size_t, const rtc::PacketTime&>
497 SignalReadPacket;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000498
499 sigslot::signal1<Connection*> SignalReadyToSend;
500
501 // Called when a packet is received on this connection.
502 void OnReadPacket(const char* data, size_t size,
503 const rtc::PacketTime& packet_time);
504
505 // Called when the socket is currently able to send.
506 void OnReadyToSend();
507
508 // Called when a connection is determined to be no longer useful to us. We
509 // still keep it around in case the other side wants to use it. But we can
510 // safely stop pinging on it and we can allow it to time out if the other
511 // side stops using it as well.
512 bool pruned() const { return pruned_; }
513 void Prune();
514
515 bool use_candidate_attr() const { return use_candidate_attr_; }
516 void set_use_candidate_attr(bool enable);
517
honghaiz5a3acd82015-08-20 15:53:17 -0700518 bool nominated() const { return nominated_; }
519 void set_nominated(bool nominated) { nominated_ = nominated; }
520
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000521 void set_remote_ice_mode(IceMode mode) {
522 remote_ice_mode_ = mode;
523 }
524
honghaiz34b11eb2016-03-16 08:55:44 -0700525 void set_receiving_timeout(int64_t receiving_timeout_ms) {
Peter Thatcher04ac81f2015-09-21 11:48:28 -0700526 receiving_timeout_ = receiving_timeout_ms;
527 }
528
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000529 // Makes the connection go away.
530 void Destroy();
531
deadbeef376e1232015-11-25 09:00:08 -0800532 // Makes the connection go away, in a failed state.
533 void FailAndDestroy();
534
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000535 // Checks that the state of this connection is up-to-date. The argument is
536 // the current time, which is compared against various timeouts.
honghaiz34b11eb2016-03-16 08:55:44 -0700537 void UpdateState(int64_t now);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000538
539 // Called when this connection should try checking writability again.
honghaiz34b11eb2016-03-16 08:55:44 -0700540 int64_t last_ping_sent() const { return last_ping_sent_; }
541 void Ping(int64_t now);
Peter Thatcher1fe120a2015-06-10 11:33:17 -0700542 void ReceivedPingResponse();
honghaiz34b11eb2016-03-16 08:55:44 -0700543 int64_t last_ping_response_received() const {
Honghai Zhang381b4212015-12-04 12:24:03 -0800544 return last_ping_response_received_;
545 }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000546
547 // Called whenever a valid ping is received on this connection. This is
548 // public because the connection intercepts the first ping for us.
honghaiz34b11eb2016-03-16 08:55:44 -0700549 int64_t last_ping_received() const { return last_ping_received_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000550 void ReceivedPing();
honghaiz9b5ee9c2015-11-11 13:19:17 -0800551 // Handles the binding request; sends a response if this is a valid request.
552 void HandleBindingRequest(IceMessage* msg);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000553
554 // Debugging description of this connection
guoweis@webrtc.org8c9ff202014-12-04 07:56:02 +0000555 std::string ToDebugId() const;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000556 std::string ToString() const;
557 std::string ToSensitiveString() const;
Peter Thatcher1cf6f812015-05-15 10:40:45 -0700558 // Prints pings_since_last_response_ into a string.
559 void PrintPingsSinceLastResponse(std::string* pings, size_t max);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000560
561 bool reported() const { return reported_; }
562 void set_reported(bool reported) { reported_ = reported;}
563
honghaiz5a3acd82015-08-20 15:53:17 -0700564 // This signal will be fired if this connection is nominated by the
565 // controlling side.
566 sigslot::signal1<Connection*> SignalNominated;
Peter Thatcher54360512015-07-08 11:08:35 -0700567
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000568 // Invoked when Connection receives STUN error response with 487 code.
569 void HandleRoleConflictFromPeer();
570
571 State state() const { return state_; }
572
573 IceMode remote_ice_mode() const { return remote_ice_mode_; }
574
honghaize1a0c942016-02-16 14:54:56 -0800575 uint32_t ComputeNetworkCost() const;
576
Taylor Brandstetter0a1bc532016-04-19 18:03:26 -0700577 // Update the ICE password and/or generation of the remote candidate if a
578 // ufrag in |remote_ice_parameters| matches the candidate's ufrag, and the
579 // candidate's password and/or ufrag has not been set.
580 // |remote_ice_parameters| should be a list of known ICE parameters ordered
581 // by generation.
582 void MaybeSetRemoteIceCredentialsAndGeneration(const std::string& ice_ufrag,
583 const std::string& ice_pwd,
584 int generation);
jiayl@webrtc.orgdacdd942015-01-23 17:33:34 +0000585
586 // If |remote_candidate_| is peer reflexive and is equivalent to
587 // |new_candidate| except the type, update |remote_candidate_| to
588 // |new_candidate|.
589 void MaybeUpdatePeerReflexiveCandidate(const Candidate& new_candidate);
590
Peter Thatcher54360512015-07-08 11:08:35 -0700591 // Returns the last received time of any data, stun request, or stun
592 // response in milliseconds
honghaiz34b11eb2016-03-16 08:55:44 -0700593 int64_t last_received() const;
Peter Thatcher54360512015-07-08 11:08:35 -0700594
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000595 protected:
Guo-wei Shiehbe508a12015-04-06 12:48:47 -0700596 enum { MSG_DELETE = 0, MSG_FIRST_AVAILABLE };
597
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000598 // Constructs a new connection to the given remote port.
599 Connection(Port* port, size_t index, const Candidate& candidate);
600
601 // Called back when StunRequestManager has a stun packet to send
602 void OnSendStunPacket(const void* data, size_t size, StunRequest* req);
603
604 // Callbacks from ConnectionRequest
Guo-wei Shiehbe508a12015-04-06 12:48:47 -0700605 virtual void OnConnectionRequestResponse(ConnectionRequest* req,
606 StunMessage* response);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000607 void OnConnectionRequestErrorResponse(ConnectionRequest* req,
608 StunMessage* response);
609 void OnConnectionRequestTimeout(ConnectionRequest* req);
Peter Thatcher1cf6f812015-05-15 10:40:45 -0700610 void OnConnectionRequestSent(ConnectionRequest* req);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000611
612 // Changes the state and signals if necessary.
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000613 void set_write_state(WriteState value);
Peter Thatcher04ac81f2015-09-21 11:48:28 -0700614 void set_receiving(bool value);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000615 void set_state(State state);
616 void set_connected(bool value);
617
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000618 void OnMessage(rtc::Message *pmsg);
619
620 Port* port_;
621 size_t local_candidate_index_;
622 Candidate remote_candidate_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000623 WriteState write_state_;
Peter Thatcher04ac81f2015-09-21 11:48:28 -0700624 bool receiving_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000625 bool connected_;
626 bool pruned_;
627 // By default |use_candidate_attr_| flag will be true,
honghaiz5a3acd82015-08-20 15:53:17 -0700628 // as we will be using aggressive nomination.
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000629 // But when peer is ice-lite, this flag "must" be initialized to false and
630 // turn on when connection becomes "best connection".
631 bool use_candidate_attr_;
honghaiz5a3acd82015-08-20 15:53:17 -0700632 // Whether this connection has been nominated by the controlling side via
633 // the use_candidate attribute.
634 bool nominated_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000635 IceMode remote_ice_mode_;
636 StunRequestManager requests_;
honghaiz34b11eb2016-03-16 08:55:44 -0700637 int rtt_;
638 int64_t last_ping_sent_; // last time we sent a ping to the other side
639 int64_t last_ping_received_; // last time we received a ping from the other
640 // side
641 int64_t last_data_received_;
642 int64_t last_ping_response_received_;
Peter Thatcher1cf6f812015-05-15 10:40:45 -0700643 std::vector<SentPing> pings_since_last_response_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000644
645 rtc::RateTracker recv_rate_tracker_;
646 rtc::RateTracker send_rate_tracker_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200647 uint32_t sent_packets_discarded_;
648 uint32_t sent_packets_total_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000649
650 private:
651 void MaybeAddPrflxCandidate(ConnectionRequest* request,
652 StunMessage* response);
653
654 bool reported_;
655 State state_;
Peter Thatcher04ac81f2015-09-21 11:48:28 -0700656 // Time duration to switch from receiving to not receiving.
honghaiz34b11eb2016-03-16 08:55:44 -0700657 int receiving_timeout_;
658 int64_t time_created_ms_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000659
660 friend class Port;
661 friend class ConnectionRequest;
662};
663
deadbeef376e1232015-11-25 09:00:08 -0800664// ProxyConnection defers all the interesting work to the port.
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000665class ProxyConnection : public Connection {
666 public:
deadbeef376e1232015-11-25 09:00:08 -0800667 ProxyConnection(Port* port, size_t index, const Candidate& remote_candidate);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000668
deadbeef376e1232015-11-25 09:00:08 -0800669 int Send(const void* data,
670 size_t size,
671 const rtc::PacketOptions& options) override;
672 int GetError() override { return error_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000673
674 private:
deadbeef376e1232015-11-25 09:00:08 -0800675 int error_ = 0;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000676};
677
678} // namespace cricket
679
680#endif // WEBRTC_P2P_BASE_PORT_H_