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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"
20#include "webrtc/p2p/base/packetsocketfactory.h"
21#include "webrtc/p2p/base/portinterface.h"
22#include "webrtc/p2p/base/stun.h"
23#include "webrtc/p2p/base/stunrequest.h"
24#include "webrtc/p2p/base/transport.h"
25#include "webrtc/base/asyncpacketsocket.h"
26#include "webrtc/base/network.h"
27#include "webrtc/base/proxyinfo.h"
28#include "webrtc/base/ratetracker.h"
29#include "webrtc/base/sigslot.h"
30#include "webrtc/base/socketaddress.h"
31#include "webrtc/base/thread.h"
32
33namespace cricket {
34
35class Connection;
36class ConnectionRequest;
37
38extern const char LOCAL_PORT_TYPE[];
39extern const char STUN_PORT_TYPE[];
40extern const char PRFLX_PORT_TYPE[];
41extern const char RELAY_PORT_TYPE[];
42
43extern const char UDP_PROTOCOL_NAME[];
44extern const char TCP_PROTOCOL_NAME[];
45extern const char SSLTCP_PROTOCOL_NAME[];
46
47// RFC 6544, TCP candidate encoding rules.
48extern const int DISCARD_PORT;
49extern const char TCPTYPE_ACTIVE_STR[];
50extern const char TCPTYPE_PASSIVE_STR[];
51extern const char TCPTYPE_SIMOPEN_STR[];
52
Honghai Zhang2b342bf2015-09-30 09:51:58 -070053// The minimum time we will wait before destroying a connection after creating
54// it.
Peter Boström0c4e06b2015-10-07 12:23:21 +020055const uint32_t MIN_CONNECTION_LIFETIME = 10 * 1000; // 10 seconds.
Peter Thatcher04ac81f2015-09-21 11:48:28 -070056
57// The timeout duration when a connection does not receive anything.
Peter Boström0c4e06b2015-10-07 12:23:21 +020058const uint32_t WEAK_CONNECTION_RECEIVE_TIMEOUT = 2500; // 2.5 seconds
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000059
60// The length of time we wait before timing out writability on a connection.
Peter Boström0c4e06b2015-10-07 12:23:21 +020061const uint32_t CONNECTION_WRITE_TIMEOUT = 15 * 1000; // 15 seconds
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000062
63// The length of time we wait before we become unwritable.
Peter Boström0c4e06b2015-10-07 12:23:21 +020064const uint32_t CONNECTION_WRITE_CONNECT_TIMEOUT = 5 * 1000; // 5 seconds
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000065
66// The number of pings that must fail to respond before we become unwritable.
Peter Boström0c4e06b2015-10-07 12:23:21 +020067const uint32_t CONNECTION_WRITE_CONNECT_FAILURES = 5;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000068
69// This is the length of time that we wait for a ping response to come back.
70const int CONNECTION_RESPONSE_TIMEOUT = 5 * 1000; // 5 seconds
71
72enum RelayType {
73 RELAY_GTURN, // Legacy google relay service.
74 RELAY_TURN // Standard (TURN) relay service.
75};
76
77enum IcePriorityValue {
78 // The reason we are choosing Relay preference 2 is because, we can run
79 // Relay from client to server on UDP/TCP/TLS. To distinguish the transport
80 // protocol, we prefer UDP over TCP over TLS.
81 // For UDP ICE_TYPE_PREFERENCE_RELAY will be 2.
82 // For TCP ICE_TYPE_PREFERENCE_RELAY will be 1.
83 // For TLS ICE_TYPE_PREFERENCE_RELAY will be 0.
84 // Check turnport.cc for setting these values.
85 ICE_TYPE_PREFERENCE_RELAY = 2,
86 ICE_TYPE_PREFERENCE_HOST_TCP = 90,
87 ICE_TYPE_PREFERENCE_SRFLX = 100,
88 ICE_TYPE_PREFERENCE_PRFLX = 110,
89 ICE_TYPE_PREFERENCE_HOST = 126
90};
91
92const char* ProtoToString(ProtocolType proto);
93bool StringToProto(const char* value, ProtocolType* proto);
94
95struct ProtocolAddress {
96 rtc::SocketAddress address;
97 ProtocolType proto;
98 bool secure;
99
100 ProtocolAddress(const rtc::SocketAddress& a, ProtocolType p)
101 : address(a), proto(p), secure(false) { }
102 ProtocolAddress(const rtc::SocketAddress& a, ProtocolType p, bool sec)
103 : address(a), proto(p), secure(sec) { }
104};
105
106typedef std::set<rtc::SocketAddress> ServerAddresses;
107
108// Represents a local communication mechanism that can be used to create
109// connections to similar mechanisms of the other client. Subclasses of this
110// one add support for specific mechanisms like local UDP ports.
111class Port : public PortInterface, public rtc::MessageHandler,
112 public sigslot::has_slots<> {
113 public:
pkasting@chromium.org332331f2014-11-06 20:19:22 +0000114 Port(rtc::Thread* thread,
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000115 rtc::PacketSocketFactory* factory,
pkasting@chromium.org332331f2014-11-06 20:19:22 +0000116 rtc::Network* network,
117 const rtc::IPAddress& ip,
118 const std::string& username_fragment,
119 const std::string& password);
120 Port(rtc::Thread* thread,
121 const std::string& type,
122 rtc::PacketSocketFactory* factory,
123 rtc::Network* network,
124 const rtc::IPAddress& ip,
Peter Boström0c4e06b2015-10-07 12:23:21 +0200125 uint16_t min_port,
126 uint16_t max_port,
pkasting@chromium.org332331f2014-11-06 20:19:22 +0000127 const std::string& username_fragment,
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000128 const std::string& password);
129 virtual ~Port();
130
131 virtual const std::string& Type() const { return type_; }
132 virtual rtc::Network* Network() const { return network_; }
133
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000134 // Methods to set/get ICE role and tiebreaker values.
135 IceRole GetIceRole() const { return ice_role_; }
136 void SetIceRole(IceRole role) { ice_role_ = role; }
137
Peter Boström0c4e06b2015-10-07 12:23:21 +0200138 void SetIceTiebreaker(uint64_t tiebreaker) { tiebreaker_ = tiebreaker; }
139 uint64_t IceTiebreaker() const { return tiebreaker_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000140
141 virtual bool SharedSocket() const { return shared_socket_; }
142 void ResetSharedSocket() { shared_socket_ = false; }
143
144 // The thread on which this port performs its I/O.
145 rtc::Thread* thread() { return thread_; }
146
147 // The factory used to create the sockets of this port.
148 rtc::PacketSocketFactory* socket_factory() const { return factory_; }
149 void set_socket_factory(rtc::PacketSocketFactory* factory) {
150 factory_ = factory;
151 }
152
153 // For debugging purposes.
154 const std::string& content_name() const { return content_name_; }
155 void set_content_name(const std::string& content_name) {
156 content_name_ = content_name;
157 }
158
159 int component() const { return component_; }
160 void set_component(int component) { component_ = component; }
161
162 bool send_retransmit_count_attribute() const {
163 return send_retransmit_count_attribute_;
164 }
165 void set_send_retransmit_count_attribute(bool enable) {
166 send_retransmit_count_attribute_ = enable;
167 }
168
169 // Identifies the generation that this port was created in.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200170 uint32_t generation() { return generation_; }
171 void set_generation(uint32_t generation) { generation_ = generation; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000172
173 // ICE requires a single username/password per content/media line. So the
174 // |ice_username_fragment_| of the ports that belongs to the same content will
175 // be the same. However this causes a small complication with our relay
176 // server, which expects different username for RTP and RTCP.
177 //
178 // To resolve this problem, we implemented the username_fragment(),
179 // which returns a different username (calculated from
180 // |ice_username_fragment_|) for RTCP in the case of ICEPROTO_GOOGLE. And the
181 // username_fragment() simply returns |ice_username_fragment_| when running
182 // in ICEPROTO_RFC5245.
183 //
184 // As a result the ICEPROTO_GOOGLE will use different usernames for RTP and
185 // RTCP. And the ICEPROTO_RFC5245 will use same username for both RTP and
186 // RTCP.
187 const std::string username_fragment() const;
188 const std::string& password() const { return password_; }
189
190 // Fired when candidates are discovered by the port. When all candidates
191 // are discovered that belong to port SignalAddressReady is fired.
192 sigslot::signal2<Port*, const Candidate&> SignalCandidateReady;
193
194 // Provides all of the above information in one handy object.
195 virtual const std::vector<Candidate>& Candidates() const {
196 return candidates_;
197 }
198
199 // SignalPortComplete is sent when port completes the task of candidates
200 // allocation.
201 sigslot::signal1<Port*> SignalPortComplete;
202 // This signal sent when port fails to allocate candidates and this port
203 // can't be used in establishing the connections. When port is in shared mode
204 // and port fails to allocate one of the candidates, port shouldn't send
205 // this signal as other candidates might be usefull in establishing the
206 // connection.
207 sigslot::signal1<Port*> SignalPortError;
208
209 // Returns a map containing all of the connections of this port, keyed by the
210 // remote address.
211 typedef std::map<rtc::SocketAddress, Connection*> AddressMap;
212 const AddressMap& connections() { return connections_; }
213
214 // Returns the connection to the given address or NULL if none exists.
215 virtual Connection* GetConnection(
216 const rtc::SocketAddress& remote_addr);
217
218 // Called each time a connection is created.
219 sigslot::signal2<Port*, Connection*> SignalConnectionCreated;
220
221 // In a shared socket mode each port which shares the socket will decide
222 // to accept the packet based on the |remote_addr|. Currently only UDP
223 // port implemented this method.
224 // TODO(mallinath) - Make it pure virtual.
225 virtual bool HandleIncomingPacket(
226 rtc::AsyncPacketSocket* socket, const char* data, size_t size,
227 const rtc::SocketAddress& remote_addr,
228 const rtc::PacketTime& packet_time) {
229 ASSERT(false);
230 return false;
231 }
232
233 // Sends a response message (normal or error) to the given request. One of
234 // these methods should be called as a response to SignalUnknownAddress.
235 // NOTE: You MUST call CreateConnection BEFORE SendBindingResponse.
236 virtual void SendBindingResponse(StunMessage* request,
237 const rtc::SocketAddress& addr);
238 virtual void SendBindingErrorResponse(
239 StunMessage* request, const rtc::SocketAddress& addr,
240 int error_code, const std::string& reason);
241
242 void set_proxy(const std::string& user_agent,
243 const rtc::ProxyInfo& proxy) {
244 user_agent_ = user_agent;
245 proxy_ = proxy;
246 }
247 const std::string& user_agent() { return user_agent_; }
248 const rtc::ProxyInfo& proxy() { return proxy_; }
249
250 virtual void EnablePortPackets();
251
252 // Called if the port has no connections and is no longer useful.
253 void Destroy();
254
255 virtual void OnMessage(rtc::Message *pmsg);
256
257 // Debugging description of this port
258 virtual std::string ToString() const;
pthatcher@webrtc.org0ba15332015-01-10 00:47:02 +0000259 const rtc::IPAddress& ip() const { return ip_; }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200260 uint16_t min_port() { return min_port_; }
261 uint16_t max_port() { return max_port_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000262
263 // Timeout shortening function to speed up unit tests.
264 void set_timeout_delay(int delay) { timeout_delay_ = delay; }
265
266 // This method will return local and remote username fragements from the
267 // stun username attribute if present.
268 bool ParseStunUsername(const StunMessage* stun_msg,
269 std::string* local_username,
Peter Thatcher7cbd1882015-09-17 18:54:52 -0700270 std::string* remote_username) const;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000271 void CreateStunUsername(const std::string& remote_username,
272 std::string* stun_username_attr_str) const;
273
274 bool MaybeIceRoleConflict(const rtc::SocketAddress& addr,
275 IceMessage* stun_msg,
276 const std::string& remote_ufrag);
277
stefanc1aeaf02015-10-15 07:26:07 -0700278 // Called when a packet has been sent to the socket.
279 void OnSentPacket(const rtc::SentPacket& sent_packet);
280
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000281 // Called when the socket is currently able to send.
282 void OnReadyToSend();
283
284 // Called when the Connection discovers a local peer reflexive candidate.
285 // Returns the index of the new local candidate.
286 size_t AddPrflxCandidate(const Candidate& local);
287
Peter Boström0c4e06b2015-10-07 12:23:21 +0200288 void set_candidate_filter(uint32_t candidate_filter) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000289 candidate_filter_ = candidate_filter;
290 }
291
292 protected:
293 enum {
honghaizd0b31432015-09-30 12:42:17 -0700294 MSG_DEAD = 0,
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000295 MSG_FIRST_AVAILABLE
296 };
297
298 void set_type(const std::string& type) { type_ = type; }
299
300 void AddAddress(const rtc::SocketAddress& address,
301 const rtc::SocketAddress& base_address,
302 const rtc::SocketAddress& related_address,
Guo-wei Shieh3d564c12015-08-19 16:51:15 -0700303 const std::string& protocol,
304 const std::string& relay_protocol,
305 const std::string& tcptype,
306 const std::string& type,
Peter Boström0c4e06b2015-10-07 12:23:21 +0200307 uint32_t type_preference,
308 uint32_t relay_preference,
Guo-wei Shieh3d564c12015-08-19 16:51:15 -0700309 bool final);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000310
311 // Adds the given connection to the list. (Deleting removes them.)
312 void AddConnection(Connection* conn);
313
314 // Called when a packet is received from an unknown address that is not
315 // currently a connection. If this is an authenticated STUN binding request,
316 // then we will signal the client.
317 void OnReadPacket(const char* data, size_t size,
318 const rtc::SocketAddress& addr,
319 ProtocolType proto);
320
321 // If the given data comprises a complete and correct STUN message then the
322 // return value is true, otherwise false. If the message username corresponds
323 // with this port's username fragment, msg will contain the parsed STUN
324 // message. Otherwise, the function may send a STUN response internally.
325 // remote_username contains the remote fragment of the STUN username.
326 bool GetStunMessage(const char* data, size_t size,
327 const rtc::SocketAddress& addr,
328 IceMessage** out_msg, std::string* out_username);
329
330 // Checks if the address in addr is compatible with the port's ip.
331 bool IsCompatibleAddress(const rtc::SocketAddress& addr);
332
333 // Returns default DSCP value.
334 rtc::DiffServCodePoint DefaultDscpValue() const {
335 // No change from what MediaChannel set.
336 return rtc::DSCP_NO_CHANGE;
337 }
338
Peter Boström0c4e06b2015-10-07 12:23:21 +0200339 uint32_t candidate_filter() { return candidate_filter_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000340
341 private:
342 void Construct();
343 // Called when one of our connections deletes itself.
344 void OnConnectionDestroyed(Connection* conn);
345
honghaizd0b31432015-09-30 12:42:17 -0700346 // Whether this port is dead, and hence, should be destroyed on the controlled
347 // side.
348 bool dead() const {
349 return ice_role_ == ICEROLE_CONTROLLED && connections_.empty();
350 }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000351
352 rtc::Thread* thread_;
353 rtc::PacketSocketFactory* factory_;
354 std::string type_;
355 bool send_retransmit_count_attribute_;
356 rtc::Network* network_;
357 rtc::IPAddress ip_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200358 uint16_t min_port_;
359 uint16_t max_port_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000360 std::string content_name_;
361 int component_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200362 uint32_t generation_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000363 // In order to establish a connection to this Port (so that real data can be
364 // sent through), the other side must send us a STUN binding request that is
365 // authenticated with this username_fragment and password.
366 // PortAllocatorSession will provide these username_fragment and password.
367 //
368 // Note: we should always use username_fragment() instead of using
369 // |ice_username_fragment_| directly. For the details see the comment on
370 // username_fragment().
371 std::string ice_username_fragment_;
372 std::string password_;
373 std::vector<Candidate> candidates_;
374 AddressMap connections_;
375 int timeout_delay_;
376 bool enable_port_packets_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000377 IceRole ice_role_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200378 uint64_t tiebreaker_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000379 bool shared_socket_;
380 // Information to use when going through a proxy.
381 std::string user_agent_;
382 rtc::ProxyInfo proxy_;
383
384 // Candidate filter is pushed down to Port such that each Port could
385 // make its own decision on how to create candidates. For example,
386 // when IceTransportsType is set to relay, both RelayPort and
387 // TurnPort will hide raddr to avoid local address leakage.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200388 uint32_t candidate_filter_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000389
390 friend class Connection;
391};
392
393// Represents a communication link between a port on the local client and a
394// port on the remote client.
395class Connection : public rtc::MessageHandler,
396 public sigslot::has_slots<> {
397 public:
Peter Thatcher1cf6f812015-05-15 10:40:45 -0700398 struct SentPing {
Peter Boström0c4e06b2015-10-07 12:23:21 +0200399 SentPing(const std::string id, uint32_t sent_time)
400 : id(id), sent_time(sent_time) {}
Peter Thatcher1cf6f812015-05-15 10:40:45 -0700401
402 std::string id;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200403 uint32_t sent_time;
Peter Thatcher1cf6f812015-05-15 10:40:45 -0700404 };
405
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000406 // States are from RFC 5245. http://tools.ietf.org/html/rfc5245#section-5.7.4
407 enum State {
408 STATE_WAITING = 0, // Check has not been performed, Waiting pair on CL.
409 STATE_INPROGRESS, // Check has been sent, transaction is in progress.
410 STATE_SUCCEEDED, // Check already done, produced a successful result.
411 STATE_FAILED // Check for this connection failed.
412 };
413
414 virtual ~Connection();
415
416 // The local port where this connection sends and receives packets.
417 Port* port() { return port_; }
418 const Port* port() const { return port_; }
419
420 // Returns the description of the local port
421 virtual const Candidate& local_candidate() const;
422
423 // Returns the description of the remote port to which we communicate.
424 const Candidate& remote_candidate() const { return remote_candidate_; }
425
426 // Returns the pair priority.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200427 uint64_t priority() const;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000428
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000429 enum WriteState {
430 STATE_WRITABLE = 0, // we have received ping responses recently
431 STATE_WRITE_UNRELIABLE = 1, // we have had a few ping failures
432 STATE_WRITE_INIT = 2, // we have yet to receive a ping response
433 STATE_WRITE_TIMEOUT = 3, // we have had a large number of ping failures
434 };
435
436 WriteState write_state() const { return write_state_; }
437 bool writable() const { return write_state_ == STATE_WRITABLE; }
Peter Thatcher04ac81f2015-09-21 11:48:28 -0700438 bool receiving() const { return receiving_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000439
440 // Determines whether the connection has finished connecting. This can only
441 // be false for TCP connections.
442 bool connected() const { return connected_; }
Honghai Zhang2b342bf2015-09-30 09:51:58 -0700443 bool weak() const { return !(writable() && receiving() && connected()); }
444 bool active() const {
445 // TODO(honghaiz): Move from using |write_state_| to using |pruned_|.
446 return write_state_ != STATE_WRITE_TIMEOUT;
447 }
448 // A connection is dead if it can be safely deleted.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200449 bool dead(uint32_t now) const;
honghaiz89374372015-09-24 13:14:47 -0700450
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000451 // Estimate of the round-trip time over this connection.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200452 uint32_t rtt() const { return rtt_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000453
454 size_t sent_total_bytes();
455 size_t sent_bytes_second();
guoweis@webrtc.org930e0042014-11-17 19:42:14 +0000456 // Used to track how many packets are discarded in the application socket due
457 // to errors.
458 size_t sent_discarded_packets();
459 size_t sent_total_packets();
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000460 size_t recv_total_bytes();
461 size_t recv_bytes_second();
462 sigslot::signal1<Connection*> SignalStateChange;
463
464 // Sent when the connection has decided that it is no longer of value. It
465 // will delete itself immediately after this call.
466 sigslot::signal1<Connection*> SignalDestroyed;
467
468 // The connection can send and receive packets asynchronously. This matches
469 // the interface of AsyncPacketSocket, which may use UDP or TCP under the
470 // covers.
471 virtual int Send(const void* data, size_t size,
472 const rtc::PacketOptions& options) = 0;
473
474 // Error if Send() returns < 0
475 virtual int GetError() = 0;
476
Peter Thatcher04ac81f2015-09-21 11:48:28 -0700477 sigslot::signal4<Connection*, const char*, size_t, const rtc::PacketTime&>
478 SignalReadPacket;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000479
480 sigslot::signal1<Connection*> SignalReadyToSend;
481
482 // Called when a packet is received on this connection.
483 void OnReadPacket(const char* data, size_t size,
484 const rtc::PacketTime& packet_time);
485
486 // Called when the socket is currently able to send.
487 void OnReadyToSend();
488
489 // Called when a connection is determined to be no longer useful to us. We
490 // still keep it around in case the other side wants to use it. But we can
491 // safely stop pinging on it and we can allow it to time out if the other
492 // side stops using it as well.
493 bool pruned() const { return pruned_; }
494 void Prune();
495
496 bool use_candidate_attr() const { return use_candidate_attr_; }
497 void set_use_candidate_attr(bool enable);
498
honghaiz5a3acd82015-08-20 15:53:17 -0700499 bool nominated() const { return nominated_; }
500 void set_nominated(bool nominated) { nominated_ = nominated; }
501
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000502 void set_remote_ice_mode(IceMode mode) {
503 remote_ice_mode_ = mode;
504 }
505
Peter Boström0c4e06b2015-10-07 12:23:21 +0200506 void set_receiving_timeout(uint32_t receiving_timeout_ms) {
Peter Thatcher04ac81f2015-09-21 11:48:28 -0700507 receiving_timeout_ = receiving_timeout_ms;
508 }
509
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000510 // Makes the connection go away.
511 void Destroy();
512
513 // Checks that the state of this connection is up-to-date. The argument is
514 // the current time, which is compared against various timeouts.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200515 void UpdateState(uint32_t now);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000516
517 // Called when this connection should try checking writability again.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200518 uint32_t last_ping_sent() const { return last_ping_sent_; }
519 void Ping(uint32_t now);
Peter Thatcher1fe120a2015-06-10 11:33:17 -0700520 void ReceivedPingResponse();
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000521
522 // Called whenever a valid ping is received on this connection. This is
523 // public because the connection intercepts the first ping for us.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200524 uint32_t last_ping_received() const { return last_ping_received_; }
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000525 void ReceivedPing();
526
527 // Debugging description of this connection
guoweis@webrtc.org8c9ff202014-12-04 07:56:02 +0000528 std::string ToDebugId() const;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000529 std::string ToString() const;
530 std::string ToSensitiveString() const;
Peter Thatcher1cf6f812015-05-15 10:40:45 -0700531 // Prints pings_since_last_response_ into a string.
532 void PrintPingsSinceLastResponse(std::string* pings, size_t max);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000533
534 bool reported() const { return reported_; }
535 void set_reported(bool reported) { reported_ = reported;}
536
honghaiz5a3acd82015-08-20 15:53:17 -0700537 // This signal will be fired if this connection is nominated by the
538 // controlling side.
539 sigslot::signal1<Connection*> SignalNominated;
Peter Thatcher54360512015-07-08 11:08:35 -0700540
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000541 // Invoked when Connection receives STUN error response with 487 code.
542 void HandleRoleConflictFromPeer();
543
544 State state() const { return state_; }
545
546 IceMode remote_ice_mode() const { return remote_ice_mode_; }
547
jiayl@webrtc.orgdacdd942015-01-23 17:33:34 +0000548 // Update the ICE password of the remote candidate if |ice_ufrag| matches
549 // the candidate's ufrag, and the candidate's passwrod has not been set.
550 void MaybeSetRemoteIceCredentials(const std::string& ice_ufrag,
551 const std::string& ice_pwd);
552
553 // If |remote_candidate_| is peer reflexive and is equivalent to
554 // |new_candidate| except the type, update |remote_candidate_| to
555 // |new_candidate|.
556 void MaybeUpdatePeerReflexiveCandidate(const Candidate& new_candidate);
557
Peter Thatcher54360512015-07-08 11:08:35 -0700558 // Returns the last received time of any data, stun request, or stun
559 // response in milliseconds
Peter Boström0c4e06b2015-10-07 12:23:21 +0200560 uint32_t last_received();
Peter Thatcher54360512015-07-08 11:08:35 -0700561
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000562 protected:
Guo-wei Shiehbe508a12015-04-06 12:48:47 -0700563 enum { MSG_DELETE = 0, MSG_FIRST_AVAILABLE };
564
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000565 // Constructs a new connection to the given remote port.
566 Connection(Port* port, size_t index, const Candidate& candidate);
567
568 // Called back when StunRequestManager has a stun packet to send
569 void OnSendStunPacket(const void* data, size_t size, StunRequest* req);
570
571 // Callbacks from ConnectionRequest
Guo-wei Shiehbe508a12015-04-06 12:48:47 -0700572 virtual void OnConnectionRequestResponse(ConnectionRequest* req,
573 StunMessage* response);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000574 void OnConnectionRequestErrorResponse(ConnectionRequest* req,
575 StunMessage* response);
576 void OnConnectionRequestTimeout(ConnectionRequest* req);
Peter Thatcher1cf6f812015-05-15 10:40:45 -0700577 void OnConnectionRequestSent(ConnectionRequest* req);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000578
579 // Changes the state and signals if necessary.
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000580 void set_write_state(WriteState value);
Peter Thatcher04ac81f2015-09-21 11:48:28 -0700581 void set_receiving(bool value);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000582 void set_state(State state);
583 void set_connected(bool value);
584
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000585 void OnMessage(rtc::Message *pmsg);
586
587 Port* port_;
588 size_t local_candidate_index_;
589 Candidate remote_candidate_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000590 WriteState write_state_;
Peter Thatcher04ac81f2015-09-21 11:48:28 -0700591 bool receiving_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000592 bool connected_;
593 bool pruned_;
594 // By default |use_candidate_attr_| flag will be true,
honghaiz5a3acd82015-08-20 15:53:17 -0700595 // as we will be using aggressive nomination.
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000596 // But when peer is ice-lite, this flag "must" be initialized to false and
597 // turn on when connection becomes "best connection".
598 bool use_candidate_attr_;
honghaiz5a3acd82015-08-20 15:53:17 -0700599 // Whether this connection has been nominated by the controlling side via
600 // the use_candidate attribute.
601 bool nominated_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000602 IceMode remote_ice_mode_;
603 StunRequestManager requests_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200604 uint32_t rtt_;
605 uint32_t last_ping_sent_; // last time we sent a ping to the other side
606 uint32_t last_ping_received_; // last time we received a ping from the other
607 // side
608 uint32_t last_data_received_;
609 uint32_t last_ping_response_received_;
Peter Thatcher1cf6f812015-05-15 10:40:45 -0700610 std::vector<SentPing> pings_since_last_response_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000611
612 rtc::RateTracker recv_rate_tracker_;
613 rtc::RateTracker send_rate_tracker_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200614 uint32_t sent_packets_discarded_;
615 uint32_t sent_packets_total_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000616
617 private:
618 void MaybeAddPrflxCandidate(ConnectionRequest* request,
619 StunMessage* response);
620
621 bool reported_;
622 State state_;
Peter Thatcher04ac81f2015-09-21 11:48:28 -0700623 // Time duration to switch from receiving to not receiving.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200624 uint32_t receiving_timeout_;
625 uint32_t time_created_ms_;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000626
627 friend class Port;
628 friend class ConnectionRequest;
629};
630
631// ProxyConnection defers all the interesting work to the port
632class ProxyConnection : public Connection {
633 public:
634 ProxyConnection(Port* port, size_t index, const Candidate& candidate);
635
636 virtual int Send(const void* data, size_t size,
637 const rtc::PacketOptions& options);
638 virtual int GetError() { return error_; }
639
640 private:
641 int error_;
642};
643
644} // namespace cricket
645
646#endif // WEBRTC_P2P_BASE_PORT_H_