| /* |
| * Copyright 2012 The WebRTC Project Authors. All rights reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_P2P_BASE_TURNPORT_H_ |
| #define WEBRTC_P2P_BASE_TURNPORT_H_ |
| |
| #include <stdio.h> |
| #include <list> |
| #include <set> |
| #include <string> |
| |
| #include "webrtc/p2p/base/port.h" |
| #include "webrtc/p2p/client/basicportallocator.h" |
| #include "webrtc/rtc_base/asyncinvoker.h" |
| #include "webrtc/rtc_base/asyncpacketsocket.h" |
| |
| namespace rtc { |
| class AsyncResolver; |
| class SignalThread; |
| } |
| |
| namespace cricket { |
| |
| extern const char TURN_PORT_TYPE[]; |
| class TurnAllocateRequest; |
| class TurnEntry; |
| |
| class TurnPort : public Port { |
| public: |
| enum PortState { |
| STATE_CONNECTING, // Initial state, cannot send any packets. |
| STATE_CONNECTED, // Socket connected, ready to send stun requests. |
| STATE_READY, // Received allocate success, can send any packets. |
| STATE_RECEIVEONLY, // Had REFRESH_REQUEST error, cannot send any packets. |
| STATE_DISCONNECTED, // TCP connection died, cannot send/receive any |
| // packets. |
| }; |
| // Create a TURN port using the shared UDP socket, |socket|. |
| static TurnPort* Create(rtc::Thread* thread, |
| rtc::PacketSocketFactory* factory, |
| rtc::Network* network, |
| rtc::AsyncPacketSocket* socket, |
| const std::string& username, // ice username. |
| const std::string& password, // ice password. |
| const ProtocolAddress& server_address, |
| const RelayCredentials& credentials, |
| int server_priority, |
| const std::string& origin) { |
| return new TurnPort(thread, factory, network, socket, username, password, |
| server_address, credentials, server_priority, origin); |
| } |
| |
| // Create a TURN port that will use a new socket, bound to |network| and |
| // using a port in the range between |min_port| and |max_port|. |
| static TurnPort* Create(rtc::Thread* thread, |
| rtc::PacketSocketFactory* factory, |
| rtc::Network* network, |
| uint16_t min_port, |
| uint16_t max_port, |
| const std::string& username, // ice username. |
| const std::string& password, // ice password. |
| const ProtocolAddress& server_address, |
| const RelayCredentials& credentials, |
| int server_priority, |
| const std::string& origin) { |
| return new TurnPort(thread, factory, network, min_port, max_port, username, |
| password, server_address, credentials, server_priority, |
| origin); |
| } |
| |
| virtual ~TurnPort(); |
| |
| const ProtocolAddress& server_address() const { return server_address_; } |
| // Returns an empty address if the local address has not been assigned. |
| rtc::SocketAddress GetLocalAddress() const; |
| |
| bool ready() const { return state_ == STATE_READY; } |
| bool connected() const { |
| return state_ == STATE_READY || state_ == STATE_CONNECTED; |
| } |
| const RelayCredentials& credentials() const { return credentials_; } |
| |
| virtual ProtocolType GetProtocol() const { return server_address_.proto; } |
| |
| virtual TlsCertPolicy GetTlsCertPolicy() const { return tls_cert_policy_; } |
| |
| virtual void SetTlsCertPolicy(TlsCertPolicy tls_cert_policy) { |
| tls_cert_policy_ = tls_cert_policy; |
| } |
| |
| virtual void PrepareAddress(); |
| virtual Connection* CreateConnection( |
| const Candidate& c, PortInterface::CandidateOrigin origin); |
| virtual int SendTo(const void* data, size_t size, |
| const rtc::SocketAddress& addr, |
| const rtc::PacketOptions& options, |
| bool payload); |
| virtual int SetOption(rtc::Socket::Option opt, int value); |
| virtual int GetOption(rtc::Socket::Option opt, int* value); |
| virtual int GetError(); |
| |
| virtual bool HandleIncomingPacket(rtc::AsyncPacketSocket* socket, |
| const char* data, size_t size, |
| const rtc::SocketAddress& remote_addr, |
| const rtc::PacketTime& packet_time); |
| virtual void OnReadPacket(rtc::AsyncPacketSocket* socket, |
| const char* data, size_t size, |
| const rtc::SocketAddress& remote_addr, |
| const rtc::PacketTime& packet_time); |
| |
| virtual void OnSentPacket(rtc::AsyncPacketSocket* socket, |
| const rtc::SentPacket& sent_packet); |
| virtual void OnReadyToSend(rtc::AsyncPacketSocket* socket); |
| virtual bool SupportsProtocol(const std::string& protocol) const { |
| // Turn port only connects to UDP candidates. |
| return protocol == UDP_PROTOCOL_NAME; |
| } |
| |
| void OnSocketConnect(rtc::AsyncPacketSocket* socket); |
| void OnSocketClose(rtc::AsyncPacketSocket* socket, int error); |
| |
| |
| const std::string& hash() const { return hash_; } |
| const std::string& nonce() const { return nonce_; } |
| |
| int error() const { return error_; } |
| |
| void OnAllocateMismatch(); |
| |
| rtc::AsyncPacketSocket* socket() const { |
| return socket_; |
| } |
| |
| // For testing only. |
| rtc::AsyncInvoker* invoker() { return &invoker_; } |
| |
| // Signal with resolved server address. |
| // Parameters are port, server address and resolved server address. |
| // This signal will be sent only if server address is resolved successfully. |
| sigslot::signal3<TurnPort*, |
| const rtc::SocketAddress&, |
| const rtc::SocketAddress&> SignalResolvedServerAddress; |
| |
| // All public methods/signals below are for testing only. |
| sigslot::signal2<TurnPort*, int> SignalTurnRefreshResult; |
| sigslot::signal3<TurnPort*, const rtc::SocketAddress&, int> |
| SignalCreatePermissionResult; |
| void FlushRequests(int msg_type) { request_manager_.Flush(msg_type); } |
| bool HasRequests() { return !request_manager_.empty(); } |
| void set_credentials(RelayCredentials& credentials) { |
| credentials_ = credentials; |
| } |
| // Finds the turn entry with |address| and sets its channel id. |
| // Returns true if the entry is found. |
| bool SetEntryChannelId(const rtc::SocketAddress& address, int channel_id); |
| // Visible for testing. |
| // Shuts down the turn port, usually because of some fatal errors. |
| void Close(); |
| |
| protected: |
| TurnPort(rtc::Thread* thread, |
| rtc::PacketSocketFactory* factory, |
| rtc::Network* network, |
| rtc::AsyncPacketSocket* socket, |
| const std::string& username, |
| const std::string& password, |
| const ProtocolAddress& server_address, |
| const RelayCredentials& credentials, |
| int server_priority, |
| const std::string& origin); |
| |
| TurnPort(rtc::Thread* thread, |
| rtc::PacketSocketFactory* factory, |
| rtc::Network* network, |
| uint16_t min_port, |
| uint16_t max_port, |
| const std::string& username, |
| const std::string& password, |
| const ProtocolAddress& server_address, |
| const RelayCredentials& credentials, |
| int server_priority, |
| const std::string& origin); |
| |
| private: |
| enum { |
| MSG_ALLOCATE_ERROR = MSG_FIRST_AVAILABLE, |
| MSG_ALLOCATE_MISMATCH, |
| MSG_TRY_ALTERNATE_SERVER, |
| MSG_REFRESH_ERROR |
| }; |
| |
| typedef std::list<TurnEntry*> EntryList; |
| typedef std::map<rtc::Socket::Option, int> SocketOptionsMap; |
| typedef std::set<rtc::SocketAddress> AttemptedServerSet; |
| |
| virtual void OnMessage(rtc::Message* pmsg); |
| virtual void HandleConnectionDestroyed(Connection* conn); |
| |
| bool CreateTurnClientSocket(); |
| |
| void set_nonce(const std::string& nonce) { nonce_ = nonce; } |
| void set_realm(const std::string& realm) { |
| if (realm != realm_) { |
| realm_ = realm; |
| UpdateHash(); |
| } |
| } |
| |
| void OnRefreshError(); |
| void HandleRefreshError(); |
| bool SetAlternateServer(const rtc::SocketAddress& address); |
| void ResolveTurnAddress(const rtc::SocketAddress& address); |
| void OnResolveResult(rtc::AsyncResolverInterface* resolver); |
| |
| void AddRequestAuthInfo(StunMessage* msg); |
| void OnSendStunPacket(const void* data, size_t size, StunRequest* request); |
| // Stun address from allocate success response. |
| // Currently used only for testing. |
| void OnStunAddress(const rtc::SocketAddress& address); |
| void OnAllocateSuccess(const rtc::SocketAddress& address, |
| const rtc::SocketAddress& stun_address); |
| void OnAllocateError(); |
| void OnAllocateRequestTimeout(); |
| |
| void HandleDataIndication(const char* data, size_t size, |
| const rtc::PacketTime& packet_time); |
| void HandleChannelData(int channel_id, const char* data, size_t size, |
| const rtc::PacketTime& packet_time); |
| void DispatchPacket(const char* data, size_t size, |
| const rtc::SocketAddress& remote_addr, |
| ProtocolType proto, const rtc::PacketTime& packet_time); |
| |
| bool ScheduleRefresh(int lifetime); |
| void SendRequest(StunRequest* request, int delay); |
| int Send(const void* data, size_t size, |
| const rtc::PacketOptions& options); |
| void UpdateHash(); |
| bool UpdateNonce(StunMessage* response); |
| void ResetNonce(); |
| |
| bool HasPermission(const rtc::IPAddress& ipaddr) const; |
| TurnEntry* FindEntry(const rtc::SocketAddress& address) const; |
| TurnEntry* FindEntry(int channel_id) const; |
| bool EntryExists(TurnEntry* e); |
| void CreateOrRefreshEntry(const rtc::SocketAddress& address); |
| void DestroyEntry(TurnEntry* entry); |
| // Destroys the entry only if |timestamp| matches the destruction timestamp |
| // in |entry|. |
| void DestroyEntryIfNotCancelled(TurnEntry* entry, int64_t timestamp); |
| void ScheduleEntryDestruction(TurnEntry* entry); |
| void CancelEntryDestruction(TurnEntry* entry); |
| |
| // Marks the connection with remote address |address| failed and |
| // pruned (a.k.a. write-timed-out). Returns true if a connection is found. |
| bool FailAndPruneConnection(const rtc::SocketAddress& address); |
| |
| // Reconstruct the URL of the server which the candidate is gathered from. |
| std::string ReconstructedServerUrl(); |
| |
| ProtocolAddress server_address_; |
| TlsCertPolicy tls_cert_policy_ = TlsCertPolicy::TLS_CERT_POLICY_SECURE; |
| RelayCredentials credentials_; |
| AttemptedServerSet attempted_server_addresses_; |
| |
| rtc::AsyncPacketSocket* socket_; |
| SocketOptionsMap socket_options_; |
| rtc::AsyncResolverInterface* resolver_; |
| int error_; |
| |
| StunRequestManager request_manager_; |
| std::string realm_; // From 401/438 response message. |
| std::string nonce_; // From 401/438 response message. |
| std::string hash_; // Digest of username:realm:password |
| |
| int next_channel_number_; |
| EntryList entries_; |
| |
| PortState state_; |
| // By default the value will be set to 0. This value will be used in |
| // calculating the candidate priority. |
| int server_priority_; |
| |
| // The number of retries made due to allocate mismatch error. |
| size_t allocate_mismatch_retries_; |
| |
| rtc::AsyncInvoker invoker_; |
| |
| friend class TurnEntry; |
| friend class TurnAllocateRequest; |
| friend class TurnRefreshRequest; |
| friend class TurnCreatePermissionRequest; |
| friend class TurnChannelBindRequest; |
| }; |
| |
| } // namespace cricket |
| |
| #endif // WEBRTC_P2P_BASE_TURNPORT_H_ |