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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2004--2005, Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#ifndef TALK_P2P_BASE_PORT_H_
29#define TALK_P2P_BASE_PORT_H_
30
31#include <string>
32#include <vector>
33#include <map>
34
wu@webrtc.orga9890802013-12-13 00:21:03 +000035#include "talk/base/asyncpacketsocket.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000036#include "talk/base/network.h"
37#include "talk/base/proxyinfo.h"
38#include "talk/base/ratetracker.h"
39#include "talk/base/sigslot.h"
40#include "talk/base/socketaddress.h"
41#include "talk/base/thread.h"
42#include "talk/p2p/base/candidate.h"
43#include "talk/p2p/base/packetsocketfactory.h"
44#include "talk/p2p/base/portinterface.h"
45#include "talk/p2p/base/stun.h"
46#include "talk/p2p/base/stunrequest.h"
47#include "talk/p2p/base/transport.h"
48
henrike@webrtc.org28e20752013-07-10 00:45:36 +000049namespace cricket {
50
51class Connection;
52class ConnectionRequest;
53
54extern const char LOCAL_PORT_TYPE[];
55extern const char STUN_PORT_TYPE[];
56extern const char PRFLX_PORT_TYPE[];
57extern const char RELAY_PORT_TYPE[];
58
59extern const char UDP_PROTOCOL_NAME[];
60extern const char TCP_PROTOCOL_NAME[];
61extern const char SSLTCP_PROTOCOL_NAME[];
62
63// The length of time we wait before timing out readability on a connection.
64const uint32 CONNECTION_READ_TIMEOUT = 30 * 1000; // 30 seconds
65
66// The length of time we wait before timing out writability on a connection.
67const uint32 CONNECTION_WRITE_TIMEOUT = 15 * 1000; // 15 seconds
68
69// The length of time we wait before we become unwritable.
70const uint32 CONNECTION_WRITE_CONNECT_TIMEOUT = 5 * 1000; // 5 seconds
71
72// The number of pings that must fail to respond before we become unwritable.
73const uint32 CONNECTION_WRITE_CONNECT_FAILURES = 5;
74
75// This is the length of time that we wait for a ping response to come back.
76const int CONNECTION_RESPONSE_TIMEOUT = 5 * 1000; // 5 seconds
77
78enum RelayType {
79 RELAY_GTURN, // Legacy google relay service.
80 RELAY_TURN // Standard (TURN) relay service.
81};
82
83enum IcePriorityValue {
84 // The reason we are choosing Relay preference 2 is because, we can run
85 // Relay from client to server on UDP/TCP/TLS. To distinguish the transport
86 // protocol, we prefer UDP over TCP over TLS.
87 // For UDP ICE_TYPE_PREFERENCE_RELAY will be 2.
88 // For TCP ICE_TYPE_PREFERENCE_RELAY will be 1.
89 // For TLS ICE_TYPE_PREFERENCE_RELAY will be 0.
90 // Check turnport.cc for setting these values.
91 ICE_TYPE_PREFERENCE_RELAY = 2,
92 ICE_TYPE_PREFERENCE_HOST_TCP = 90,
93 ICE_TYPE_PREFERENCE_SRFLX = 100,
94 ICE_TYPE_PREFERENCE_PRFLX = 110,
95 ICE_TYPE_PREFERENCE_HOST = 126
96};
97
98const char* ProtoToString(ProtocolType proto);
99bool StringToProto(const char* value, ProtocolType* proto);
100
101struct ProtocolAddress {
102 talk_base::SocketAddress address;
103 ProtocolType proto;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000104 bool secure;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000105
106 ProtocolAddress(const talk_base::SocketAddress& a, ProtocolType p)
wu@webrtc.org91053e72013-08-10 07:18:04 +0000107 : address(a), proto(p), secure(false) { }
108 ProtocolAddress(const talk_base::SocketAddress& a, ProtocolType p, bool sec)
109 : address(a), proto(p), secure(sec) { }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000110};
111
112// Represents a local communication mechanism that can be used to create
113// connections to similar mechanisms of the other client. Subclasses of this
114// one add support for specific mechanisms like local UDP ports.
115class Port : public PortInterface, public talk_base::MessageHandler,
116 public sigslot::has_slots<> {
117 public:
sergeyu@chromium.orga23f0ca2013-11-13 22:48:52 +0000118 Port(talk_base::Thread* thread, talk_base::PacketSocketFactory* factory,
119 talk_base::Network* network, const talk_base::IPAddress& ip,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000120 const std::string& username_fragment, const std::string& password);
121 Port(talk_base::Thread* thread, const std::string& type,
122 talk_base::PacketSocketFactory* factory,
123 talk_base::Network* network, const talk_base::IPAddress& ip,
124 int min_port, int max_port, const std::string& username_fragment,
125 const std::string& password);
126 virtual ~Port();
127
128 virtual const std::string& Type() const { return type_; }
129 virtual talk_base::Network* Network() const { return network_; }
130
131 // This method will set the flag which enables standard ICE/STUN procedures
132 // in STUN connectivity checks. Currently this method does
133 // 1. Add / Verify MI attribute in STUN binding requests.
134 // 2. Username attribute in STUN binding request will be RFRAF:LFRAG,
135 // as opposed to RFRAGLFRAG.
136 virtual void SetIceProtocolType(IceProtocolType protocol) {
137 ice_protocol_ = protocol;
138 }
139 virtual IceProtocolType IceProtocol() const { return ice_protocol_; }
140
141 // Methods to set/get ICE role and tiebreaker values.
mallinath@webrtc.orga5506692013-08-12 21:18:15 +0000142 IceRole GetIceRole() const { return ice_role_; }
143 void SetIceRole(IceRole role) { ice_role_ = role; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000144
mallinath@webrtc.orga5506692013-08-12 21:18:15 +0000145 void SetIceTiebreaker(uint64 tiebreaker) { tiebreaker_ = tiebreaker; }
146 uint64 IceTiebreaker() const { return tiebreaker_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000147
148 virtual bool SharedSocket() const { return shared_socket_; }
149
150 // The thread on which this port performs its I/O.
151 talk_base::Thread* thread() { return thread_; }
152
153 // The factory used to create the sockets of this port.
154 talk_base::PacketSocketFactory* socket_factory() const { return factory_; }
155 void set_socket_factory(talk_base::PacketSocketFactory* factory) {
156 factory_ = factory;
157 }
158
159 // For debugging purposes.
160 const std::string& content_name() const { return content_name_; }
161 void set_content_name(const std::string& content_name) {
162 content_name_ = content_name;
163 }
164
165 int component() const { return component_; }
166 void set_component(int component) { component_ = component; }
167
168 bool send_retransmit_count_attribute() const {
169 return send_retransmit_count_attribute_;
170 }
171 void set_send_retransmit_count_attribute(bool enable) {
172 send_retransmit_count_attribute_ = enable;
173 }
174
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000175 // Identifies the generation that this port was created in.
176 uint32 generation() { return generation_; }
177 void set_generation(uint32 generation) { generation_ = generation; }
178
179 // ICE requires a single username/password per content/media line. So the
180 // |ice_username_fragment_| of the ports that belongs to the same content will
181 // be the same. However this causes a small complication with our relay
182 // server, which expects different username for RTP and RTCP.
183 //
184 // To resolve this problem, we implemented the username_fragment(),
185 // which returns a different username (calculated from
186 // |ice_username_fragment_|) for RTCP in the case of ICEPROTO_GOOGLE. And the
187 // username_fragment() simply returns |ice_username_fragment_| when running
188 // in ICEPROTO_RFC5245.
189 //
190 // As a result the ICEPROTO_GOOGLE will use different usernames for RTP and
191 // RTCP. And the ICEPROTO_RFC5245 will use same username for both RTP and
192 // RTCP.
193 const std::string username_fragment() const;
194 const std::string& password() const { return password_; }
195
196 // Fired when candidates are discovered by the port. When all candidates
197 // are discovered that belong to port SignalAddressReady is fired.
198 sigslot::signal2<Port*, const Candidate&> SignalCandidateReady;
199
200 // Provides all of the above information in one handy object.
201 virtual const std::vector<Candidate>& Candidates() const {
202 return candidates_;
203 }
204
205 // SignalPortComplete is sent when port completes the task of candidates
206 // allocation.
207 sigslot::signal1<Port*> SignalPortComplete;
208 // This signal sent when port fails to allocate candidates and this port
209 // can't be used in establishing the connections. When port is in shared mode
210 // and port fails to allocate one of the candidates, port shouldn't send
211 // this signal as other candidates might be usefull in establishing the
212 // connection.
213 sigslot::signal1<Port*> SignalPortError;
214
215 // Returns a map containing all of the connections of this port, keyed by the
216 // remote address.
217 typedef std::map<talk_base::SocketAddress, Connection*> AddressMap;
218 const AddressMap& connections() { return connections_; }
219
220 // Returns the connection to the given address or NULL if none exists.
221 virtual Connection* GetConnection(
222 const talk_base::SocketAddress& remote_addr);
223
224 // Called each time a connection is created.
225 sigslot::signal2<Port*, Connection*> SignalConnectionCreated;
226
227 // In a shared socket mode each port which shares the socket will decide
228 // to accept the packet based on the |remote_addr|. Currently only UDP
229 // port implemented this method.
230 // TODO(mallinath) - Make it pure virtual.
231 virtual bool HandleIncomingPacket(
232 talk_base::AsyncPacketSocket* socket, const char* data, size_t size,
wu@webrtc.orga9890802013-12-13 00:21:03 +0000233 const talk_base::SocketAddress& remote_addr,
234 const talk_base::PacketTime& packet_time) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000235 ASSERT(false);
236 return false;
237 }
238
239 // Sends a response message (normal or error) to the given request. One of
240 // these methods should be called as a response to SignalUnknownAddress.
241 // NOTE: You MUST call CreateConnection BEFORE SendBindingResponse.
242 virtual void SendBindingResponse(StunMessage* request,
243 const talk_base::SocketAddress& addr);
244 virtual void SendBindingErrorResponse(
245 StunMessage* request, const talk_base::SocketAddress& addr,
246 int error_code, const std::string& reason);
247
248 void set_proxy(const std::string& user_agent,
249 const talk_base::ProxyInfo& proxy) {
250 user_agent_ = user_agent;
251 proxy_ = proxy;
252 }
253 const std::string& user_agent() { return user_agent_; }
254 const talk_base::ProxyInfo& proxy() { return proxy_; }
255
256 virtual void EnablePortPackets();
257
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000258 // Called if the port has no connections and is no longer useful.
259 void Destroy();
260
261 virtual void OnMessage(talk_base::Message *pmsg);
262
263 // Debugging description of this port
264 virtual std::string ToString() const;
265 talk_base::IPAddress& ip() { return ip_; }
266 int min_port() { return min_port_; }
267 int max_port() { return max_port_; }
268
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000269 // Timeout shortening function to speed up unit tests.
270 void set_timeout_delay(int delay) { timeout_delay_ = delay; }
271
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000272 // This method will return local and remote username fragements from the
273 // stun username attribute if present.
274 bool ParseStunUsername(const StunMessage* stun_msg,
275 std::string* local_username,
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000276 std::string* remote_username,
277 IceProtocolType* remote_protocol_type) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000278 void CreateStunUsername(const std::string& remote_username,
279 std::string* stun_username_attr_str) const;
280
281 bool MaybeIceRoleConflict(const talk_base::SocketAddress& addr,
282 IceMessage* stun_msg,
283 const std::string& remote_ufrag);
284
285 // Called when the socket is currently able to send.
286 void OnReadyToSend();
287
288 // Called when the Connection discovers a local peer reflexive candidate.
289 // Returns the index of the new local candidate.
290 size_t AddPrflxCandidate(const Candidate& local);
291
292 // Returns if RFC 5245 ICE protocol is used.
293 bool IsStandardIce() const;
294
295 // Returns if Google ICE protocol is used.
296 bool IsGoogleIce() const;
297
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000298 // Returns if Hybrid ICE protocol is used.
299 bool IsHybridIce() const;
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000300
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000301 protected:
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000302 enum {
303 MSG_CHECKTIMEOUT = 0,
304 MSG_FIRST_AVAILABLE
305 };
306
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000307 void set_type(const std::string& type) { type_ = type; }
308 // Fills in the local address of the port.
309 void AddAddress(const talk_base::SocketAddress& address,
310 const talk_base::SocketAddress& base_address,
buildbot@webrtc.orgf875f152014-04-14 16:06:21 +0000311 const talk_base::SocketAddress& related_address,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000312 const std::string& protocol, const std::string& type,
313 uint32 type_preference, bool final);
314
315 // Adds the given connection to the list. (Deleting removes them.)
316 void AddConnection(Connection* conn);
317
318 // Called when a packet is received from an unknown address that is not
319 // currently a connection. If this is an authenticated STUN binding request,
320 // then we will signal the client.
321 void OnReadPacket(const char* data, size_t size,
322 const talk_base::SocketAddress& addr,
323 ProtocolType proto);
324
325 // If the given data comprises a complete and correct STUN message then the
326 // return value is true, otherwise false. If the message username corresponds
327 // with this port's username fragment, msg will contain the parsed STUN
328 // message. Otherwise, the function may send a STUN response internally.
329 // remote_username contains the remote fragment of the STUN username.
330 bool GetStunMessage(const char* data, size_t size,
331 const talk_base::SocketAddress& addr,
332 IceMessage** out_msg, std::string* out_username);
333
334 // Checks if the address in addr is compatible with the port's ip.
335 bool IsCompatibleAddress(const talk_base::SocketAddress& addr);
336
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000337 // Returns default DSCP value.
338 talk_base::DiffServCodePoint DefaultDscpValue() const {
339 // No change from what MediaChannel set.
340 return talk_base::DSCP_NO_CHANGE;
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000341 }
342
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000343 private:
344 void Construct();
345 // Called when one of our connections deletes itself.
346 void OnConnectionDestroyed(Connection* conn);
347
348 // Checks if this port is useless, and hence, should be destroyed.
349 void CheckTimeout();
350
351 talk_base::Thread* thread_;
352 talk_base::PacketSocketFactory* factory_;
353 std::string type_;
354 bool send_retransmit_count_attribute_;
355 talk_base::Network* network_;
356 talk_base::IPAddress ip_;
357 int min_port_;
358 int max_port_;
359 std::string content_name_;
360 int component_;
361 uint32 generation_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000362 // In order to establish a connection to this Port (so that real data can be
363 // sent through), the other side must send us a STUN binding request that is
364 // authenticated with this username_fragment and password.
365 // PortAllocatorSession will provide these username_fragment and password.
366 //
367 // Note: we should always use username_fragment() instead of using
368 // |ice_username_fragment_| directly. For the details see the comment on
369 // username_fragment().
370 std::string ice_username_fragment_;
371 std::string password_;
372 std::vector<Candidate> candidates_;
373 AddressMap connections_;
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000374 int timeout_delay_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000375 bool enable_port_packets_;
376 IceProtocolType ice_protocol_;
mallinath@webrtc.orga5506692013-08-12 21:18:15 +0000377 IceRole ice_role_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000378 uint64 tiebreaker_;
379 bool shared_socket_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000380 // Information to use when going through a proxy.
381 std::string user_agent_;
382 talk_base::ProxyInfo proxy_;
383
384 friend class Connection;
385};
386
387// Represents a communication link between a port on the local client and a
388// port on the remote client.
389class Connection : public talk_base::MessageHandler,
390 public sigslot::has_slots<> {
391 public:
392 // States are from RFC 5245. http://tools.ietf.org/html/rfc5245#section-5.7.4
393 enum State {
394 STATE_WAITING = 0, // Check has not been performed, Waiting pair on CL.
395 STATE_INPROGRESS, // Check has been sent, transaction is in progress.
396 STATE_SUCCEEDED, // Check already done, produced a successful result.
397 STATE_FAILED // Check for this connection failed.
398 };
399
400 virtual ~Connection();
401
402 // The local port where this connection sends and receives packets.
403 Port* port() { return port_; }
404 const Port* port() const { return port_; }
405
406 // Returns the description of the local port
407 virtual const Candidate& local_candidate() const;
408
409 // Returns the description of the remote port to which we communicate.
410 const Candidate& remote_candidate() const { return remote_candidate_; }
411
412 // Returns the pair priority.
413 uint64 priority() const;
414
415 enum ReadState {
416 STATE_READ_INIT = 0, // we have yet to receive a ping
417 STATE_READABLE = 1, // we have received pings recently
418 STATE_READ_TIMEOUT = 2, // we haven't received pings in a while
419 };
420
421 ReadState read_state() const { return read_state_; }
422 bool readable() const { return read_state_ == STATE_READABLE; }
423
424 enum WriteState {
425 STATE_WRITABLE = 0, // we have received ping responses recently
426 STATE_WRITE_UNRELIABLE = 1, // we have had a few ping failures
427 STATE_WRITE_INIT = 2, // we have yet to receive a ping response
428 STATE_WRITE_TIMEOUT = 3, // we have had a large number of ping failures
429 };
430
431 WriteState write_state() const { return write_state_; }
432 bool writable() const { return write_state_ == STATE_WRITABLE; }
433
434 // Determines whether the connection has finished connecting. This can only
435 // be false for TCP connections.
436 bool connected() const { return connected_; }
437
438 // Estimate of the round-trip time over this connection.
439 uint32 rtt() const { return rtt_; }
440
441 size_t sent_total_bytes();
442 size_t sent_bytes_second();
443 size_t recv_total_bytes();
444 size_t recv_bytes_second();
445 sigslot::signal1<Connection*> SignalStateChange;
446
447 // Sent when the connection has decided that it is no longer of value. It
448 // will delete itself immediately after this call.
449 sigslot::signal1<Connection*> SignalDestroyed;
450
451 // The connection can send and receive packets asynchronously. This matches
452 // the interface of AsyncPacketSocket, which may use UDP or TCP under the
453 // covers.
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000454 virtual int Send(const void* data, size_t size,
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000455 const talk_base::PacketOptions& options) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000456
457 // Error if Send() returns < 0
458 virtual int GetError() = 0;
459
wu@webrtc.orga9890802013-12-13 00:21:03 +0000460 sigslot::signal4<Connection*, const char*, size_t,
461 const talk_base::PacketTime&> SignalReadPacket;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000462
463 sigslot::signal1<Connection*> SignalReadyToSend;
464
465 // Called when a packet is received on this connection.
wu@webrtc.orga9890802013-12-13 00:21:03 +0000466 void OnReadPacket(const char* data, size_t size,
467 const talk_base::PacketTime& packet_time);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000468
469 // Called when the socket is currently able to send.
470 void OnReadyToSend();
471
472 // Called when a connection is determined to be no longer useful to us. We
473 // still keep it around in case the other side wants to use it. But we can
474 // safely stop pinging on it and we can allow it to time out if the other
475 // side stops using it as well.
476 bool pruned() const { return pruned_; }
477 void Prune();
478
479 bool use_candidate_attr() const { return use_candidate_attr_; }
480 void set_use_candidate_attr(bool enable);
481
482 void set_remote_ice_mode(IceMode mode) {
483 remote_ice_mode_ = mode;
484 }
485
486 // Makes the connection go away.
487 void Destroy();
488
489 // Checks that the state of this connection is up-to-date. The argument is
490 // the current time, which is compared against various timeouts.
491 void UpdateState(uint32 now);
492
493 // Called when this connection should try checking writability again.
494 uint32 last_ping_sent() const { return last_ping_sent_; }
495 void Ping(uint32 now);
496
497 // Called whenever a valid ping is received on this connection. This is
498 // public because the connection intercepts the first ping for us.
499 uint32 last_ping_received() const { return last_ping_received_; }
500 void ReceivedPing();
501
502 // Debugging description of this connection
503 std::string ToString() const;
504 std::string ToSensitiveString() const;
505
506 bool reported() const { return reported_; }
507 void set_reported(bool reported) { reported_ = reported;}
508
509 // This flag will be set if this connection is the chosen one for media
510 // transmission. This connection will send STUN ping with USE-CANDIDATE
511 // attribute.
512 sigslot::signal1<Connection*> SignalUseCandidate;
513 // Invoked when Connection receives STUN error response with 487 code.
514 void HandleRoleConflictFromPeer();
515
516 State state() const { return state_; }
517
518 IceMode remote_ice_mode() const { return remote_ice_mode_; }
519
520 protected:
521 // Constructs a new connection to the given remote port.
522 Connection(Port* port, size_t index, const Candidate& candidate);
523
524 // Called back when StunRequestManager has a stun packet to send
525 void OnSendStunPacket(const void* data, size_t size, StunRequest* req);
526
527 // Callbacks from ConnectionRequest
528 void OnConnectionRequestResponse(ConnectionRequest* req,
529 StunMessage* response);
530 void OnConnectionRequestErrorResponse(ConnectionRequest* req,
531 StunMessage* response);
532 void OnConnectionRequestTimeout(ConnectionRequest* req);
533
534 // Changes the state and signals if necessary.
535 void set_read_state(ReadState value);
536 void set_write_state(WriteState value);
537 void set_state(State state);
538 void set_connected(bool value);
539
540 // Checks if this connection is useless, and hence, should be destroyed.
541 void CheckTimeout();
542
543 void OnMessage(talk_base::Message *pmsg);
544
545 Port* port_;
546 size_t local_candidate_index_;
547 Candidate remote_candidate_;
548 ReadState read_state_;
549 WriteState write_state_;
550 bool connected_;
551 bool pruned_;
552 // By default |use_candidate_attr_| flag will be true,
553 // as we will be using agrressive nomination.
554 // But when peer is ice-lite, this flag "must" be initialized to false and
555 // turn on when connection becomes "best connection".
556 bool use_candidate_attr_;
557 IceMode remote_ice_mode_;
558 StunRequestManager requests_;
559 uint32 rtt_;
560 uint32 last_ping_sent_; // last time we sent a ping to the other side
561 uint32 last_ping_received_; // last time we received a ping from the other
562 // side
563 uint32 last_data_received_;
564 uint32 last_ping_response_received_;
565 std::vector<uint32> pings_since_last_response_;
566
567 talk_base::RateTracker recv_rate_tracker_;
568 talk_base::RateTracker send_rate_tracker_;
569
570 private:
571 void MaybeAddPrflxCandidate(ConnectionRequest* request,
572 StunMessage* response);
573
574 bool reported_;
575 State state_;
576
577 friend class Port;
578 friend class ConnectionRequest;
579};
580
581// ProxyConnection defers all the interesting work to the port
582class ProxyConnection : public Connection {
583 public:
584 ProxyConnection(Port* port, size_t index, const Candidate& candidate);
585
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000586 virtual int Send(const void* data, size_t size,
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000587 const talk_base::PacketOptions& options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000588 virtual int GetError() { return error_; }
589
590 private:
591 int error_;
592};
593
594} // namespace cricket
595
596#endif // TALK_P2P_BASE_PORT_H_