b6173abe59
As part of implementing IceTransportsType constraint, we should hide the raddr which is the mapped address to prevent local address leakage. BUG=1179 R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7484 4adac7df-926f-26a2-2b94-8c16560cd09d
620 lines
23 KiB
C++
620 lines
23 KiB
C++
/*
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* libjingle
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* Copyright 2004--2005, Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef TALK_P2P_BASE_PORT_H_
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#define TALK_P2P_BASE_PORT_H_
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#include <map>
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#include <set>
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#include <string>
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#include <vector>
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#include "talk/p2p/base/candidate.h"
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#include "talk/p2p/base/packetsocketfactory.h"
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#include "talk/p2p/base/portinterface.h"
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#include "talk/p2p/base/stun.h"
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#include "talk/p2p/base/stunrequest.h"
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#include "talk/p2p/base/transport.h"
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#include "webrtc/base/asyncpacketsocket.h"
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#include "webrtc/base/network.h"
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#include "webrtc/base/proxyinfo.h"
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#include "webrtc/base/ratetracker.h"
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#include "webrtc/base/sigslot.h"
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#include "webrtc/base/socketaddress.h"
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#include "webrtc/base/thread.h"
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namespace cricket {
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class Connection;
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class ConnectionRequest;
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extern const char LOCAL_PORT_TYPE[];
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extern const char STUN_PORT_TYPE[];
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extern const char PRFLX_PORT_TYPE[];
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extern const char RELAY_PORT_TYPE[];
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extern const char UDP_PROTOCOL_NAME[];
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extern const char TCP_PROTOCOL_NAME[];
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extern const char SSLTCP_PROTOCOL_NAME[];
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// RFC 6544, TCP candidate encoding rules.
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extern const int DISCARD_PORT;
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extern const char TCPTYPE_ACTIVE_STR[];
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extern const char TCPTYPE_PASSIVE_STR[];
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extern const char TCPTYPE_SIMOPEN_STR[];
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// The length of time we wait before timing out readability on a connection.
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const uint32 CONNECTION_READ_TIMEOUT = 30 * 1000; // 30 seconds
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// The length of time we wait before timing out writability on a connection.
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const uint32 CONNECTION_WRITE_TIMEOUT = 15 * 1000; // 15 seconds
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// The length of time we wait before we become unwritable.
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const uint32 CONNECTION_WRITE_CONNECT_TIMEOUT = 5 * 1000; // 5 seconds
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// The number of pings that must fail to respond before we become unwritable.
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const uint32 CONNECTION_WRITE_CONNECT_FAILURES = 5;
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// This is the length of time that we wait for a ping response to come back.
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const int CONNECTION_RESPONSE_TIMEOUT = 5 * 1000; // 5 seconds
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enum RelayType {
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RELAY_GTURN, // Legacy google relay service.
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RELAY_TURN // Standard (TURN) relay service.
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};
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enum IcePriorityValue {
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// The reason we are choosing Relay preference 2 is because, we can run
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// Relay from client to server on UDP/TCP/TLS. To distinguish the transport
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// protocol, we prefer UDP over TCP over TLS.
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// For UDP ICE_TYPE_PREFERENCE_RELAY will be 2.
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// For TCP ICE_TYPE_PREFERENCE_RELAY will be 1.
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// For TLS ICE_TYPE_PREFERENCE_RELAY will be 0.
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// Check turnport.cc for setting these values.
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ICE_TYPE_PREFERENCE_RELAY = 2,
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ICE_TYPE_PREFERENCE_HOST_TCP = 90,
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ICE_TYPE_PREFERENCE_SRFLX = 100,
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ICE_TYPE_PREFERENCE_PRFLX = 110,
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ICE_TYPE_PREFERENCE_HOST = 126
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};
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const char* ProtoToString(ProtocolType proto);
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bool StringToProto(const char* value, ProtocolType* proto);
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struct ProtocolAddress {
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rtc::SocketAddress address;
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ProtocolType proto;
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bool secure;
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ProtocolAddress(const rtc::SocketAddress& a, ProtocolType p)
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: address(a), proto(p), secure(false) { }
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ProtocolAddress(const rtc::SocketAddress& a, ProtocolType p, bool sec)
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: address(a), proto(p), secure(sec) { }
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};
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typedef std::set<rtc::SocketAddress> ServerAddresses;
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// Represents a local communication mechanism that can be used to create
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// connections to similar mechanisms of the other client. Subclasses of this
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// one add support for specific mechanisms like local UDP ports.
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class Port : public PortInterface, public rtc::MessageHandler,
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public sigslot::has_slots<> {
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public:
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Port(rtc::Thread* thread, rtc::PacketSocketFactory* factory,
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rtc::Network* network, const rtc::IPAddress& ip,
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const std::string& username_fragment, const std::string& password);
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Port(rtc::Thread* thread, const std::string& type,
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rtc::PacketSocketFactory* factory,
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rtc::Network* network, const rtc::IPAddress& ip,
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int min_port, int max_port, const std::string& username_fragment,
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const std::string& password);
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virtual ~Port();
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virtual const std::string& Type() const { return type_; }
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virtual rtc::Network* Network() const { return network_; }
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// This method will set the flag which enables standard ICE/STUN procedures
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// in STUN connectivity checks. Currently this method does
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// 1. Add / Verify MI attribute in STUN binding requests.
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// 2. Username attribute in STUN binding request will be RFRAF:LFRAG,
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// as opposed to RFRAGLFRAG.
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virtual void SetIceProtocolType(IceProtocolType protocol) {
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ice_protocol_ = protocol;
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}
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virtual IceProtocolType IceProtocol() const { return ice_protocol_; }
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// Methods to set/get ICE role and tiebreaker values.
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IceRole GetIceRole() const { return ice_role_; }
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void SetIceRole(IceRole role) { ice_role_ = role; }
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void SetIceTiebreaker(uint64 tiebreaker) { tiebreaker_ = tiebreaker; }
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uint64 IceTiebreaker() const { return tiebreaker_; }
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virtual bool SharedSocket() const { return shared_socket_; }
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void ResetSharedSocket() { shared_socket_ = false; }
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// The thread on which this port performs its I/O.
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rtc::Thread* thread() { return thread_; }
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// The factory used to create the sockets of this port.
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rtc::PacketSocketFactory* socket_factory() const { return factory_; }
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void set_socket_factory(rtc::PacketSocketFactory* factory) {
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factory_ = factory;
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}
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// For debugging purposes.
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const std::string& content_name() const { return content_name_; }
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void set_content_name(const std::string& content_name) {
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content_name_ = content_name;
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}
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int component() const { return component_; }
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void set_component(int component) { component_ = component; }
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bool send_retransmit_count_attribute() const {
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return send_retransmit_count_attribute_;
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}
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void set_send_retransmit_count_attribute(bool enable) {
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send_retransmit_count_attribute_ = enable;
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}
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// Identifies the generation that this port was created in.
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uint32 generation() { return generation_; }
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void set_generation(uint32 generation) { generation_ = generation; }
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// ICE requires a single username/password per content/media line. So the
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// |ice_username_fragment_| of the ports that belongs to the same content will
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// be the same. However this causes a small complication with our relay
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// server, which expects different username for RTP and RTCP.
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//
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// To resolve this problem, we implemented the username_fragment(),
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// which returns a different username (calculated from
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// |ice_username_fragment_|) for RTCP in the case of ICEPROTO_GOOGLE. And the
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// username_fragment() simply returns |ice_username_fragment_| when running
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// in ICEPROTO_RFC5245.
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//
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// As a result the ICEPROTO_GOOGLE will use different usernames for RTP and
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// RTCP. And the ICEPROTO_RFC5245 will use same username for both RTP and
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// RTCP.
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const std::string username_fragment() const;
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const std::string& password() const { return password_; }
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// Fired when candidates are discovered by the port. When all candidates
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// are discovered that belong to port SignalAddressReady is fired.
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sigslot::signal2<Port*, const Candidate&> SignalCandidateReady;
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// Provides all of the above information in one handy object.
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virtual const std::vector<Candidate>& Candidates() const {
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return candidates_;
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}
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// SignalPortComplete is sent when port completes the task of candidates
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// allocation.
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sigslot::signal1<Port*> SignalPortComplete;
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// This signal sent when port fails to allocate candidates and this port
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// can't be used in establishing the connections. When port is in shared mode
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// and port fails to allocate one of the candidates, port shouldn't send
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// this signal as other candidates might be usefull in establishing the
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// connection.
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sigslot::signal1<Port*> SignalPortError;
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// Returns a map containing all of the connections of this port, keyed by the
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// remote address.
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typedef std::map<rtc::SocketAddress, Connection*> AddressMap;
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const AddressMap& connections() { return connections_; }
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// Returns the connection to the given address or NULL if none exists.
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virtual Connection* GetConnection(
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const rtc::SocketAddress& remote_addr);
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// Called each time a connection is created.
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sigslot::signal2<Port*, Connection*> SignalConnectionCreated;
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// In a shared socket mode each port which shares the socket will decide
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// to accept the packet based on the |remote_addr|. Currently only UDP
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// port implemented this method.
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// TODO(mallinath) - Make it pure virtual.
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virtual bool HandleIncomingPacket(
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rtc::AsyncPacketSocket* socket, const char* data, size_t size,
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const rtc::SocketAddress& remote_addr,
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const rtc::PacketTime& packet_time) {
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ASSERT(false);
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return false;
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}
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// Sends a response message (normal or error) to the given request. One of
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// these methods should be called as a response to SignalUnknownAddress.
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// NOTE: You MUST call CreateConnection BEFORE SendBindingResponse.
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virtual void SendBindingResponse(StunMessage* request,
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const rtc::SocketAddress& addr);
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virtual void SendBindingErrorResponse(
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StunMessage* request, const rtc::SocketAddress& addr,
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int error_code, const std::string& reason);
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void set_proxy(const std::string& user_agent,
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const rtc::ProxyInfo& proxy) {
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user_agent_ = user_agent;
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proxy_ = proxy;
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}
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const std::string& user_agent() { return user_agent_; }
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const rtc::ProxyInfo& proxy() { return proxy_; }
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virtual void EnablePortPackets();
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// Called if the port has no connections and is no longer useful.
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void Destroy();
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virtual void OnMessage(rtc::Message *pmsg);
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// Debugging description of this port
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virtual std::string ToString() const;
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rtc::IPAddress& ip() { return ip_; }
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int min_port() { return min_port_; }
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int max_port() { return max_port_; }
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// Timeout shortening function to speed up unit tests.
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void set_timeout_delay(int delay) { timeout_delay_ = delay; }
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// This method will return local and remote username fragements from the
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// stun username attribute if present.
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bool ParseStunUsername(const StunMessage* stun_msg,
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std::string* local_username,
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std::string* remote_username,
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IceProtocolType* remote_protocol_type) const;
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void CreateStunUsername(const std::string& remote_username,
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std::string* stun_username_attr_str) const;
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bool MaybeIceRoleConflict(const rtc::SocketAddress& addr,
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IceMessage* stun_msg,
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const std::string& remote_ufrag);
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// Called when the socket is currently able to send.
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void OnReadyToSend();
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// Called when the Connection discovers a local peer reflexive candidate.
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// Returns the index of the new local candidate.
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size_t AddPrflxCandidate(const Candidate& local);
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// Returns if RFC 5245 ICE protocol is used.
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bool IsStandardIce() const;
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// Returns if Google ICE protocol is used.
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bool IsGoogleIce() const;
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// Returns if Hybrid ICE protocol is used.
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bool IsHybridIce() const;
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void set_candidate_filter(uint32 candidate_filter) {
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candidate_filter_ = candidate_filter;
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}
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protected:
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enum {
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MSG_CHECKTIMEOUT = 0,
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MSG_FIRST_AVAILABLE
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};
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void set_type(const std::string& type) { type_ = type; }
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void AddAddress(const rtc::SocketAddress& address,
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const rtc::SocketAddress& base_address,
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const rtc::SocketAddress& related_address,
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const std::string& protocol, const std::string& tcptype,
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const std::string& type, uint32 type_preference,
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uint32 relay_preference, bool final);
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// Adds the given connection to the list. (Deleting removes them.)
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void AddConnection(Connection* conn);
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// Called when a packet is received from an unknown address that is not
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// currently a connection. If this is an authenticated STUN binding request,
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// then we will signal the client.
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void OnReadPacket(const char* data, size_t size,
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const rtc::SocketAddress& addr,
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ProtocolType proto);
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// If the given data comprises a complete and correct STUN message then the
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// return value is true, otherwise false. If the message username corresponds
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// with this port's username fragment, msg will contain the parsed STUN
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// message. Otherwise, the function may send a STUN response internally.
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// remote_username contains the remote fragment of the STUN username.
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bool GetStunMessage(const char* data, size_t size,
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const rtc::SocketAddress& addr,
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IceMessage** out_msg, std::string* out_username);
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// Checks if the address in addr is compatible with the port's ip.
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bool IsCompatibleAddress(const rtc::SocketAddress& addr);
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// Returns default DSCP value.
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rtc::DiffServCodePoint DefaultDscpValue() const {
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// No change from what MediaChannel set.
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return rtc::DSCP_NO_CHANGE;
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}
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uint32 candidate_filter() { return candidate_filter_; }
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private:
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void Construct();
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// Called when one of our connections deletes itself.
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void OnConnectionDestroyed(Connection* conn);
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// Checks if this port is useless, and hence, should be destroyed.
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void CheckTimeout();
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rtc::Thread* thread_;
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rtc::PacketSocketFactory* factory_;
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std::string type_;
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bool send_retransmit_count_attribute_;
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rtc::Network* network_;
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rtc::IPAddress ip_;
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int min_port_;
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int max_port_;
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std::string content_name_;
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int component_;
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uint32 generation_;
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// In order to establish a connection to this Port (so that real data can be
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// sent through), the other side must send us a STUN binding request that is
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// authenticated with this username_fragment and password.
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// PortAllocatorSession will provide these username_fragment and password.
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//
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// Note: we should always use username_fragment() instead of using
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// |ice_username_fragment_| directly. For the details see the comment on
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// username_fragment().
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std::string ice_username_fragment_;
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std::string password_;
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std::vector<Candidate> candidates_;
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AddressMap connections_;
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int timeout_delay_;
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bool enable_port_packets_;
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IceProtocolType ice_protocol_;
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IceRole ice_role_;
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uint64 tiebreaker_;
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bool shared_socket_;
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// Information to use when going through a proxy.
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std::string user_agent_;
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rtc::ProxyInfo proxy_;
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// Candidate filter is pushed down to Port such that each Port could
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// make its own decision on how to create candidates. For example,
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// when IceTransportsType is set to relay, both RelayPort and
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// TurnPort will hide raddr to avoid local address leakage.
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uint32 candidate_filter_;
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friend class Connection;
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};
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// Represents a communication link between a port on the local client and a
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// port on the remote client.
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class Connection : public rtc::MessageHandler,
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public sigslot::has_slots<> {
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public:
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// States are from RFC 5245. http://tools.ietf.org/html/rfc5245#section-5.7.4
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enum State {
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STATE_WAITING = 0, // Check has not been performed, Waiting pair on CL.
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STATE_INPROGRESS, // Check has been sent, transaction is in progress.
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STATE_SUCCEEDED, // Check already done, produced a successful result.
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STATE_FAILED // Check for this connection failed.
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};
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virtual ~Connection();
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// The local port where this connection sends and receives packets.
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Port* port() { return port_; }
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const Port* port() const { return port_; }
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// Returns the description of the local port
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virtual const Candidate& local_candidate() const;
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// Returns the description of the remote port to which we communicate.
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const Candidate& remote_candidate() const { return remote_candidate_; }
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// Returns the pair priority.
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uint64 priority() const;
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enum ReadState {
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STATE_READ_INIT = 0, // we have yet to receive a ping
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STATE_READABLE = 1, // we have received pings recently
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STATE_READ_TIMEOUT = 2, // we haven't received pings in a while
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};
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ReadState read_state() const { return read_state_; }
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bool readable() const { return read_state_ == STATE_READABLE; }
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enum WriteState {
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STATE_WRITABLE = 0, // we have received ping responses recently
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STATE_WRITE_UNRELIABLE = 1, // we have had a few ping failures
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STATE_WRITE_INIT = 2, // we have yet to receive a ping response
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STATE_WRITE_TIMEOUT = 3, // we have had a large number of ping failures
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};
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WriteState write_state() const { return write_state_; }
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bool writable() const { return write_state_ == STATE_WRITABLE; }
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// Determines whether the connection has finished connecting. This can only
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// be false for TCP connections.
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bool connected() const { return connected_; }
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// Estimate of the round-trip time over this connection.
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uint32 rtt() const { return rtt_; }
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size_t sent_total_bytes();
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size_t sent_bytes_second();
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size_t recv_total_bytes();
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size_t recv_bytes_second();
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sigslot::signal1<Connection*> SignalStateChange;
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// Sent when the connection has decided that it is no longer of value. It
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// will delete itself immediately after this call.
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sigslot::signal1<Connection*> SignalDestroyed;
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// The connection can send and receive packets asynchronously. This matches
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// the interface of AsyncPacketSocket, which may use UDP or TCP under the
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// covers.
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virtual int Send(const void* data, size_t size,
|
|
const rtc::PacketOptions& options) = 0;
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|
|
|
// Error if Send() returns < 0
|
|
virtual int GetError() = 0;
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|
|
|
sigslot::signal4<Connection*, const char*, size_t,
|
|
const rtc::PacketTime&> SignalReadPacket;
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|
|
|
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);
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|
|
|
// Called when the socket is currently able to send.
|
|
void OnReadyToSend();
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|
|
|
// 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.
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|
bool pruned() const { return pruned_; }
|
|
void Prune();
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|
|
|
bool use_candidate_attr() const { return use_candidate_attr_; }
|
|
void set_use_candidate_attr(bool enable);
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|
|
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void set_remote_ice_mode(IceMode mode) {
|
|
remote_ice_mode_ = mode;
|
|
}
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|
|
|
// Makes the connection go away.
|
|
void Destroy();
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|
|
|
// 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);
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|
|
|
// Called when this connection should try checking writability again.
|
|
uint32 last_ping_sent() const { return last_ping_sent_; }
|
|
void Ping(uint32 now);
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|
|
|
// 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 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_;
|
|
|
|
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 // TALK_P2P_BASE_PORT_H_
|