c8554be6dd
Wrap the socket in an SSL adapter, then simply call StartSSL() on the SSLAdapter instance. Cloned from: https://webrtc-codereview.appspot.com/21799004/ R=juberti@chromium.org, juberti@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/14059004 Patch from Manish Jethani <manish.jethani@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@6852 4adac7df-926f-26a2-2b94-8c16560cd09d
222 lines
7.2 KiB
C++
222 lines
7.2 KiB
C++
/*
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* libjingle
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* Copyright 2011, 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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#include "talk/p2p/base/basicpacketsocketfactory.h"
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#include "webrtc/base/asyncudpsocket.h"
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#include "webrtc/base/asynctcpsocket.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/base/nethelpers.h"
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#include "webrtc/base/physicalsocketserver.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/socketadapters.h"
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#include "webrtc/base/ssladapter.h"
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#include "webrtc/base/thread.h"
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#include "talk/p2p/base/asyncstuntcpsocket.h"
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#include "talk/p2p/base/stun.h"
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namespace rtc {
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BasicPacketSocketFactory::BasicPacketSocketFactory()
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: thread_(Thread::Current()),
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socket_factory_(NULL) {
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}
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BasicPacketSocketFactory::BasicPacketSocketFactory(Thread* thread)
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: thread_(thread),
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socket_factory_(NULL) {
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}
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BasicPacketSocketFactory::BasicPacketSocketFactory(
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SocketFactory* socket_factory)
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: thread_(NULL),
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socket_factory_(socket_factory) {
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}
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BasicPacketSocketFactory::~BasicPacketSocketFactory() {
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}
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AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket(
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const SocketAddress& address, int min_port, int max_port) {
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// UDP sockets are simple.
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rtc::AsyncSocket* socket =
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socket_factory()->CreateAsyncSocket(
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address.family(), SOCK_DGRAM);
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if (!socket) {
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return NULL;
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}
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if (BindSocket(socket, address, min_port, max_port) < 0) {
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LOG(LS_ERROR) << "UDP bind failed with error "
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<< socket->GetError();
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delete socket;
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return NULL;
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}
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return new rtc::AsyncUDPSocket(socket);
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}
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AsyncPacketSocket* BasicPacketSocketFactory::CreateServerTcpSocket(
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const SocketAddress& local_address, int min_port, int max_port, int opts) {
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// Fail if TLS is required.
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if (opts & PacketSocketFactory::OPT_TLS) {
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LOG(LS_ERROR) << "TLS support currently is not available.";
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return NULL;
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}
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rtc::AsyncSocket* socket =
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socket_factory()->CreateAsyncSocket(local_address.family(),
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SOCK_STREAM);
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if (!socket) {
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return NULL;
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}
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if (BindSocket(socket, local_address, min_port, max_port) < 0) {
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LOG(LS_ERROR) << "TCP bind failed with error "
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<< socket->GetError();
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delete socket;
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return NULL;
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}
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// If using SSLTCP, wrap the TCP socket in a pseudo-SSL socket.
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if (opts & PacketSocketFactory::OPT_SSLTCP) {
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ASSERT(!(opts & PacketSocketFactory::OPT_TLS));
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socket = new rtc::AsyncSSLSocket(socket);
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}
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// Set TCP_NODELAY (via OPT_NODELAY) for improved performance.
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// See http://go/gtalktcpnodelayexperiment
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socket->SetOption(rtc::Socket::OPT_NODELAY, 1);
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if (opts & PacketSocketFactory::OPT_STUN)
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return new cricket::AsyncStunTCPSocket(socket, true);
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return new rtc::AsyncTCPSocket(socket, true);
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}
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AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket(
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const SocketAddress& local_address, const SocketAddress& remote_address,
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const ProxyInfo& proxy_info, const std::string& user_agent, int opts) {
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rtc::AsyncSocket* socket =
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socket_factory()->CreateAsyncSocket(local_address.family(), SOCK_STREAM);
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if (!socket) {
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return NULL;
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}
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if (BindSocket(socket, local_address, 0, 0) < 0) {
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LOG(LS_ERROR) << "TCP bind failed with error "
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<< socket->GetError();
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delete socket;
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return NULL;
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}
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// If using a proxy, wrap the socket in a proxy socket.
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if (proxy_info.type == rtc::PROXY_SOCKS5) {
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socket = new rtc::AsyncSocksProxySocket(
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socket, proxy_info.address, proxy_info.username, proxy_info.password);
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} else if (proxy_info.type == rtc::PROXY_HTTPS) {
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socket = new rtc::AsyncHttpsProxySocket(
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socket, user_agent, proxy_info.address,
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proxy_info.username, proxy_info.password);
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}
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// If using TLS, wrap the socket in an SSL adapter.
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if (opts & PacketSocketFactory::OPT_TLS) {
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ASSERT(!(opts & PacketSocketFactory::OPT_SSLTCP));
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rtc::SSLAdapter* ssl_adapter = rtc::SSLAdapter::Create(socket);
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if (!ssl_adapter) {
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return NULL;
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}
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socket = ssl_adapter;
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if (ssl_adapter->StartSSL(remote_address.hostname().c_str(), false) != 0) {
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delete ssl_adapter;
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return NULL;
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}
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// If using SSLTCP, wrap the TCP socket in a pseudo-SSL socket.
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} else if (opts & PacketSocketFactory::OPT_SSLTCP) {
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ASSERT(!(opts & PacketSocketFactory::OPT_TLS));
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socket = new rtc::AsyncSSLSocket(socket);
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}
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if (socket->Connect(remote_address) < 0) {
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LOG(LS_ERROR) << "TCP connect failed with error "
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<< socket->GetError();
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delete socket;
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return NULL;
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}
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// Finally, wrap that socket in a TCP or STUN TCP packet socket.
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AsyncPacketSocket* tcp_socket;
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if (opts & PacketSocketFactory::OPT_STUN) {
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tcp_socket = new cricket::AsyncStunTCPSocket(socket, false);
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} else {
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tcp_socket = new rtc::AsyncTCPSocket(socket, false);
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}
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// Set TCP_NODELAY (via OPT_NODELAY) for improved performance.
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// See http://go/gtalktcpnodelayexperiment
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tcp_socket->SetOption(rtc::Socket::OPT_NODELAY, 1);
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return tcp_socket;
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}
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AsyncResolverInterface* BasicPacketSocketFactory::CreateAsyncResolver() {
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return new rtc::AsyncResolver();
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}
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int BasicPacketSocketFactory::BindSocket(
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AsyncSocket* socket, const SocketAddress& local_address,
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int min_port, int max_port) {
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int ret = -1;
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if (min_port == 0 && max_port == 0) {
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// If there's no port range, let the OS pick a port for us.
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ret = socket->Bind(local_address);
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} else {
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// Otherwise, try to find a port in the provided range.
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for (int port = min_port; ret < 0 && port <= max_port; ++port) {
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ret = socket->Bind(rtc::SocketAddress(local_address.ipaddr(),
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port));
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}
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}
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return ret;
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}
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SocketFactory* BasicPacketSocketFactory::socket_factory() {
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if (thread_) {
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ASSERT(thread_ == Thread::Current());
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return thread_->socketserver();
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} else {
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return socket_factory_;
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}
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}
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} // namespace rtc
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