Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS. Maintaining deprecated
backwards compatible support for "OPT_SSLTCP".
Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.
PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.
TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.
Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.
For security reasons we ensure that:
1) The policy is always explicitly initialized to secure.
2) API users have to explicitly integrate with the feature to
use it, and will otherwise get no change in behavior.
3) The feature is not immediately exposed in non-native
contexts. For example, disabling of certificate validation
is not implemented via URI parsing since this would
immediately allow it to be used from a web page.
This is a second attempt of https://codereview.webrtc.org/2557803002/
which was rolled back in https://codereview.webrtc.org/2590153002/
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2594623002
Cr-Commit-Position: refs/heads/master@{#15967}
215 lines
6.5 KiB
C++
215 lines
6.5 KiB
C++
/*
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* Copyright 2011 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/p2p/base/basicpacketsocketfactory.h"
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#include <string>
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#include "webrtc/p2p/base/asyncstuntcpsocket.h"
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#include "webrtc/p2p/base/stun.h"
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#include "webrtc/base/asynctcpsocket.h"
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#include "webrtc/base/asyncudpsocket.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/socketadapters.h"
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#include "webrtc/base/ssladapter.h"
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#include "webrtc/base/thread.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,
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uint16_t min_port,
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uint16_t max_port) {
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// UDP sockets are simple.
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AsyncSocket* socket =
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socket_factory()->CreateAsyncSocket(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 AsyncUDPSocket(socket);
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}
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AsyncPacketSocket* BasicPacketSocketFactory::CreateServerTcpSocket(
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const SocketAddress& local_address,
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uint16_t min_port,
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uint16_t max_port,
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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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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, 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 fake TLS, wrap the TCP socket in a pseudo-SSL socket.
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if (opts & PacketSocketFactory::OPT_TLS_FAKE) {
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ASSERT(!(opts & PacketSocketFactory::OPT_TLS));
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socket = new 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(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 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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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 == PROXY_SOCKS5) {
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socket = new AsyncSocksProxySocket(
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socket, proxy_info.address, proxy_info.username, proxy_info.password);
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} else if (proxy_info.type == PROXY_HTTPS) {
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socket =
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new AsyncHttpsProxySocket(socket, user_agent, proxy_info.address,
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proxy_info.username, proxy_info.password);
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}
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// Assert that at most one TLS option is used.
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int tlsOpts =
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opts & (PacketSocketFactory::OPT_TLS | PacketSocketFactory::OPT_TLS_FAKE |
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PacketSocketFactory::OPT_TLS_INSECURE);
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ASSERT((tlsOpts & (tlsOpts - 1)) == 0);
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if ((tlsOpts & PacketSocketFactory::OPT_TLS) ||
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(tlsOpts & PacketSocketFactory::OPT_TLS_INSECURE)) {
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// Using TLS, wrap the socket in an SSL adapter.
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SSLAdapter* ssl_adapter = SSLAdapter::Create(socket);
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if (!ssl_adapter) {
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return NULL;
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}
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if (tlsOpts & PacketSocketFactory::OPT_TLS_INSECURE) {
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ssl_adapter->set_ignore_bad_cert(true);
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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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} else if (tlsOpts & PacketSocketFactory::OPT_TLS_FAKE) {
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// Using fake TLS, wrap the TCP socket in a pseudo-SSL socket.
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socket = new 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 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(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 AsyncResolver();
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}
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int BasicPacketSocketFactory::BindSocket(AsyncSocket* socket,
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const SocketAddress& local_address,
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uint16_t min_port,
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uint16_t 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(SocketAddress(local_address.ipaddr(), 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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