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/*
* Copyright 2011 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.
*/
#include "webrtc/p2p/base/basicpacketsocketfactory.h"
#include "webrtc/p2p/base/asyncstuntcpsocket.h"
#include "webrtc/p2p/base/stun.h"
#include "webrtc/base/asynctcpsocket.h"
#include "webrtc/base/asyncudpsocket.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/nethelpers.h"
#include "webrtc/base/physicalsocketserver.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/socketadapters.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/thread.h"
namespace rtc {
BasicPacketSocketFactory::BasicPacketSocketFactory()
: thread_(Thread::Current()),
socket_factory_(NULL) {
}
BasicPacketSocketFactory::BasicPacketSocketFactory(Thread* thread)
: thread_(thread),
socket_factory_(NULL) {
}
BasicPacketSocketFactory::BasicPacketSocketFactory(
SocketFactory* socket_factory)
: thread_(NULL),
socket_factory_(socket_factory) {
}
BasicPacketSocketFactory::~BasicPacketSocketFactory() {
}
AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket(
const SocketAddress& address,
uint16_t min_port,
uint16_t max_port) {
// UDP sockets are simple.
rtc::AsyncSocket* socket =
socket_factory()->CreateAsyncSocket(
address.family(), SOCK_DGRAM);
if (!socket) {
return NULL;
}
if (BindSocket(socket, address, min_port, max_port) < 0) {
LOG(LS_ERROR) << "UDP bind failed with error "
<< socket->GetError();
delete socket;
return NULL;
}
return new rtc::AsyncUDPSocket(socket);
}
AsyncPacketSocket* BasicPacketSocketFactory::CreateServerTcpSocket(
const SocketAddress& local_address,
uint16_t min_port,
uint16_t max_port,
int opts) {
// Fail if TLS is required.
if (opts & PacketSocketFactory::OPT_TLS) {
LOG(LS_ERROR) << "TLS support currently is not available.";
return NULL;
}
rtc::AsyncSocket* socket =
socket_factory()->CreateAsyncSocket(local_address.family(),
SOCK_STREAM);
if (!socket) {
return NULL;
}
if (BindSocket(socket, local_address, min_port, max_port) < 0) {
LOG(LS_ERROR) << "TCP bind failed with error "
<< socket->GetError();
delete socket;
return NULL;
}
// If using SSLTCP, wrap the TCP socket in a pseudo-SSL socket.
if (opts & PacketSocketFactory::OPT_SSLTCP) {
ASSERT(!(opts & PacketSocketFactory::OPT_TLS));
socket = new rtc::AsyncSSLSocket(socket);
}
// Set TCP_NODELAY (via OPT_NODELAY) for improved performance.
// See http://go/gtalktcpnodelayexperiment
socket->SetOption(rtc::Socket::OPT_NODELAY, 1);
if (opts & PacketSocketFactory::OPT_STUN)
return new cricket::AsyncStunTCPSocket(socket, true);
return new rtc::AsyncTCPSocket(socket, true);
}
AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket(
const SocketAddress& local_address, const SocketAddress& remote_address,
const ProxyInfo& proxy_info, const std::string& user_agent, int opts) {
rtc::AsyncSocket* socket =
socket_factory()->CreateAsyncSocket(local_address.family(), SOCK_STREAM);
if (!socket) {
return NULL;
}
if (BindSocket(socket, local_address, 0, 0) < 0) {
LOG(LS_ERROR) << "TCP bind failed with error "
<< socket->GetError();
delete socket;
return NULL;
}
// If using a proxy, wrap the socket in a proxy socket.
if (proxy_info.type == rtc::PROXY_SOCKS5) {
socket = new rtc::AsyncSocksProxySocket(
socket, proxy_info.address, proxy_info.username, proxy_info.password);
} else if (proxy_info.type == rtc::PROXY_HTTPS) {
socket = new rtc::AsyncHttpsProxySocket(
socket, user_agent, proxy_info.address,
proxy_info.username, proxy_info.password);
}
// If using TLS, wrap the socket in an SSL adapter.
if (opts & PacketSocketFactory::OPT_TLS) {
ASSERT(!(opts & PacketSocketFactory::OPT_SSLTCP));
rtc::SSLAdapter* ssl_adapter = rtc::SSLAdapter::Create(socket);
if (!ssl_adapter) {
return NULL;
}
socket = ssl_adapter;
if (ssl_adapter->StartSSL(remote_address.hostname().c_str(), false) != 0) {
delete ssl_adapter;
return NULL;
}
// If using SSLTCP, wrap the TCP socket in a pseudo-SSL socket.
} else if (opts & PacketSocketFactory::OPT_SSLTCP) {
ASSERT(!(opts & PacketSocketFactory::OPT_TLS));
socket = new rtc::AsyncSSLSocket(socket);
}
if (socket->Connect(remote_address) < 0) {
LOG(LS_ERROR) << "TCP connect failed with error "
<< socket->GetError();
delete socket;
return NULL;
}
// Finally, wrap that socket in a TCP or STUN TCP packet socket.
AsyncPacketSocket* tcp_socket;
if (opts & PacketSocketFactory::OPT_STUN) {
tcp_socket = new cricket::AsyncStunTCPSocket(socket, false);
} else {
tcp_socket = new rtc::AsyncTCPSocket(socket, false);
}
// Set TCP_NODELAY (via OPT_NODELAY) for improved performance.
// See http://go/gtalktcpnodelayexperiment
tcp_socket->SetOption(rtc::Socket::OPT_NODELAY, 1);
return tcp_socket;
}
AsyncResolverInterface* BasicPacketSocketFactory::CreateAsyncResolver() {
return new rtc::AsyncResolver();
}
int BasicPacketSocketFactory::BindSocket(AsyncSocket* socket,
const SocketAddress& local_address,
uint16_t min_port,
uint16_t max_port) {
int ret = -1;
if (min_port == 0 && max_port == 0) {
// If there's no port range, let the OS pick a port for us.
ret = socket->Bind(local_address);
} else {
// Otherwise, try to find a port in the provided range.
for (int port = min_port; ret < 0 && port <= max_port; ++port) {
ret = socket->Bind(rtc::SocketAddress(local_address.ipaddr(),
port));
}
}
return ret;
}
SocketFactory* BasicPacketSocketFactory::socket_factory() {
if (thread_) {
ASSERT(thread_ == Thread::Current());
return thread_->socketserver();
} else {
return socket_factory_;
}
}
} // namespace rtc
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