webrtc_m130/p2p/base/tcpport.h

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/*
* Copyright 2004 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_TCPPORT_H_
#define WEBRTC_P2P_BASE_TCPPORT_H_
#include <list>
#include <memory>
#include <string>
#include "webrtc/p2p/base/port.h"
#include "webrtc/rtc_base/asyncpacketsocket.h"
namespace cricket {
class TCPConnection;
// Communicates using a local TCP port.
//
// This class is designed to allow subclasses to take advantage of the
// connection management provided by this class. A subclass should take of all
// packet sending and preparation, but when a packet is received, it should
// call this TCPPort::OnReadPacket (3 arg) to dispatch to a connection.
class TCPPort : public Port {
public:
static TCPPort* Create(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,
bool allow_listen) {
Make Port (and subclasses) fully "Network"-based, instead of IP-based. For ICE, we want sockets that are bound to specific network interfaces, rather than to specific IP addresses. So, a while ago, we added a "Network" class that gets passed into the Port constructor, in addition to the IP address as before. But we never finished the job of removing the IP address field, such that a Port only guarantees something about the network interface it's associated with, and not the specific IP address it ends up with. This CL does that, and as a consequence, if a port ends up bound to an IP address other than the "best" one (returned by Network::GetBestIP), this *won't* be treated as an error. This is relevant to Android, where even though we pass an IP address into "Bind" as a way of identifying the network, the socket actually gets bound using "android_setsocknetwork", which doesn't provide any guarantees about the IP address. So, if a network interface has multiple IPv6 addresses (for instance), we may not correctly predict the one the OS will choose, and that's ok. This CL also moves "SetAlternateLocalAddress" from VirtualSocket to VirtualSocketServer, which makes for much more readable test code. The next step, if there is one, is to pass along the Network class all the way to SocketServer::Bind. Then the socket server could do smart things with the network information. We could even stick a platform- specific network handle in the Network object, such that the socket server could use it for the binding, or for "sendmsg", for example. See bug 7026 for more context about the sendmsg idea. BUG=webrtc:7715 Review-Url: https://codereview.webrtc.org/2989303002 Cr-Commit-Position: refs/heads/master@{#19251}
2017-08-04 15:01:57 -07:00
return new TCPPort(thread, factory, network, min_port, max_port, username,
password, allow_listen);
}
~TCPPort() override;
Connection* CreateConnection(const Candidate& address,
CandidateOrigin origin) override;
void PrepareAddress() override;
int GetOption(rtc::Socket::Option opt, int* value) override;
int SetOption(rtc::Socket::Option opt, int value) override;
int GetError() override;
bool SupportsProtocol(const std::string& protocol) const override {
return protocol == TCP_PROTOCOL_NAME || protocol == SSLTCP_PROTOCOL_NAME;
}
ProtocolType GetProtocol() const override { return PROTO_TCP; }
protected:
TCPPort(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,
bool allow_listen);
// Handles sending using the local TCP socket.
int SendTo(const void* data,
size_t size,
const rtc::SocketAddress& addr,
const rtc::PacketOptions& options,
bool payload) override;
// Accepts incoming TCP connection.
void OnNewConnection(rtc::AsyncPacketSocket* socket,
rtc::AsyncPacketSocket* new_socket);
private:
struct Incoming {
rtc::SocketAddress addr;
rtc::AsyncPacketSocket* socket;
};
void TryCreateServerSocket();
rtc::AsyncPacketSocket* GetIncoming(
const rtc::SocketAddress& addr, bool remove = false);
// Receives packet signal from the local TCP Socket.
void OnReadPacket(rtc::AsyncPacketSocket* socket,
const char* data, size_t size,
const rtc::SocketAddress& remote_addr,
const rtc::PacketTime& packet_time);
void OnSentPacket(rtc::AsyncPacketSocket* socket,
const rtc::SentPacket& sent_packet) override;
void OnReadyToSend(rtc::AsyncPacketSocket* socket);
void OnAddressReady(rtc::AsyncPacketSocket* socket,
const rtc::SocketAddress& address);
// TODO: Is this still needed?
bool incoming_only_;
bool allow_listen_;
rtc::AsyncPacketSocket* socket_;
int error_;
std::list<Incoming> incoming_;
friend class TCPConnection;
};
class TCPConnection : public Connection {
public:
// Connection is outgoing unless socket is specified
TCPConnection(TCPPort* port, const Candidate& candidate,
rtc::AsyncPacketSocket* socket = 0);
~TCPConnection() override;
int Send(const void* data,
size_t size,
const rtc::PacketOptions& options) override;
int GetError() override;
rtc::AsyncPacketSocket* socket() { return socket_.get(); }
void OnMessage(rtc::Message* pmsg) override;
// Allow test cases to overwrite the default timeout period.
int reconnection_timeout() const { return reconnection_timeout_; }
void set_reconnection_timeout(int timeout_in_ms) {
reconnection_timeout_ = timeout_in_ms;
}
protected:
enum {
MSG_TCPCONNECTION_DELAYED_ONCLOSE = Connection::MSG_FIRST_AVAILABLE,
};
// Set waiting_for_stun_binding_complete_ to false to allow data packets in
// addition to what Port::OnConnectionRequestResponse does.
void OnConnectionRequestResponse(ConnectionRequest* req,
StunMessage* response) override;
private:
// Helper function to handle the case when Ping or Send fails with error
// related to socket close.
void MaybeReconnect();
void CreateOutgoingTcpSocket();
void ConnectSocketSignals(rtc::AsyncPacketSocket* socket);
void OnConnect(rtc::AsyncPacketSocket* socket);
void OnClose(rtc::AsyncPacketSocket* socket, int error);
void OnReadPacket(rtc::AsyncPacketSocket* socket,
const char* data, size_t size,
const rtc::SocketAddress& remote_addr,
const rtc::PacketTime& packet_time);
void OnReadyToSend(rtc::AsyncPacketSocket* socket);
std::unique_ptr<rtc::AsyncPacketSocket> socket_;
int error_;
bool outgoing_;
// Guard against multiple outgoing tcp connection during a reconnect.
bool connection_pending_;
// Guard against data packets sent when we reconnect a TCP connection. During
// reconnecting, when a new tcp connection has being made, we can't send data
// packets out until the STUN binding is completed (i.e. the write state is
// set to WRITABLE again by Connection::OnConnectionRequestResponse). IPC
// socket, when receiving data packets before that, will trigger OnError which
// will terminate the newly created connection.
bool pretending_to_be_writable_;
// Allow test case to overwrite the default timeout period.
int reconnection_timeout_;
friend class TCPPort;
};
} // namespace cricket
#endif // WEBRTC_P2P_BASE_TCPPORT_H_