webrtc/p2p/base/basic_packet_socket_factory.cc
Harald Alvestrand aa653c0d76 Reland "Deprecate all classes related to AsyncResolver"
This reverts commit 08d431ec34.

Reason for revert: Last (hopefully) Chrome blocker removed

Original change's description:
> Revert "Deprecate all classes related to AsyncResolver"
>
> This reverts commit 61a442809c.
>
> Reason for revert: Breaks roll into Chromium
>
> Original change's description:
> > Deprecate all classes related to AsyncResolver
> >
> > and remove internal usage.
> >
> > Bug: webrtc:12598
> > Change-Id: Ie208682bfa0163f6c7a8e805151cfbda76324496
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322860
> > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> > Auto-Submit: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#40919}
>
> Bug: webrtc:12598
> Change-Id: I8aef5e062e19a51baec75873eddfca2a10467d3c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322901
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40927}

Bug: webrtc:12598
Change-Id: I3c7b07c831eb9ff808368433d9b9ae8ec4b2afb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323720
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40944}
2023-10-17 07:08:57 +00:00

210 lines
6.9 KiB
C++

/*
* 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 "p2p/base/basic_packet_socket_factory.h"
#include <stddef.h>
#include <string>
#include "absl/memory/memory.h"
#include "api/async_dns_resolver.h"
#include "api/wrapping_async_dns_resolver.h"
#include "p2p/base/async_stun_tcp_socket.h"
#include "rtc_base/async_dns_resolver.h"
#include "rtc_base/async_tcp_socket.h"
#include "rtc_base/async_udp_socket.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/socket.h"
#include "rtc_base/socket_adapters.h"
#include "rtc_base/ssl_adapter.h"
namespace rtc {
BasicPacketSocketFactory::BasicPacketSocketFactory(
SocketFactory* socket_factory)
: socket_factory_(socket_factory) {}
BasicPacketSocketFactory::~BasicPacketSocketFactory() {}
AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket(
const SocketAddress& address,
uint16_t min_port,
uint16_t max_port) {
// UDP sockets are simple.
Socket* socket = socket_factory_->CreateSocket(address.family(), SOCK_DGRAM);
if (!socket) {
return NULL;
}
if (BindSocket(socket, address, min_port, max_port) < 0) {
RTC_LOG(LS_ERROR) << "UDP bind failed with error " << socket->GetError();
delete socket;
return NULL;
}
return new AsyncUDPSocket(socket);
}
AsyncListenSocket* 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) {
RTC_LOG(LS_ERROR) << "TLS support currently is not available.";
return NULL;
}
if (opts & PacketSocketFactory::OPT_TLS_FAKE) {
RTC_LOG(LS_ERROR) << "Fake TLS not supported.";
return NULL;
}
Socket* socket =
socket_factory_->CreateSocket(local_address.family(), SOCK_STREAM);
if (!socket) {
return NULL;
}
if (BindSocket(socket, local_address, min_port, max_port) < 0) {
RTC_LOG(LS_ERROR) << "TCP bind failed with error " << socket->GetError();
delete socket;
return NULL;
}
RTC_CHECK(!(opts & PacketSocketFactory::OPT_STUN));
return new AsyncTcpListenSocket(absl::WrapUnique(socket));
}
AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket(
const SocketAddress& local_address,
const SocketAddress& remote_address,
const ProxyInfo& proxy_info,
const std::string& user_agent,
const PacketSocketTcpOptions& tcp_options) {
Socket* socket =
socket_factory_->CreateSocket(local_address.family(), SOCK_STREAM);
if (!socket) {
return NULL;
}
if (BindSocket(socket, local_address, 0, 0) < 0) {
// Allow BindSocket to fail if we're binding to the ANY address, since this
// is mostly redundant in the first place. The socket will be bound when we
// call Connect() instead.
if (local_address.IsAnyIP()) {
RTC_LOG(LS_WARNING) << "TCP bind failed with error " << socket->GetError()
<< "; ignoring since socket is using 'any' address.";
} else {
RTC_LOG(LS_ERROR) << "TCP bind failed with error " << socket->GetError();
delete socket;
return NULL;
}
}
// Set TCP_NODELAY (via OPT_NODELAY) for improved performance; this causes
// small media packets to be sent immediately rather than being buffered up,
// reducing latency.
//
// Must be done before calling Connect, otherwise it may fail.
if (socket->SetOption(Socket::OPT_NODELAY, 1) != 0) {
RTC_LOG(LS_ERROR) << "Setting TCP_NODELAY option failed with error "
<< socket->GetError();
}
// If using a proxy, wrap the socket in a proxy socket.
if (proxy_info.type == PROXY_SOCKS5) {
socket = new AsyncSocksProxySocket(
socket, proxy_info.address, proxy_info.username, proxy_info.password);
} else if (proxy_info.type == PROXY_HTTPS) {
socket =
new AsyncHttpsProxySocket(socket, user_agent, proxy_info.address,
proxy_info.username, proxy_info.password);
}
// Assert that at most one TLS option is used.
int tlsOpts = tcp_options.opts & (PacketSocketFactory::OPT_TLS |
PacketSocketFactory::OPT_TLS_FAKE |
PacketSocketFactory::OPT_TLS_INSECURE);
RTC_DCHECK((tlsOpts & (tlsOpts - 1)) == 0);
if ((tlsOpts & PacketSocketFactory::OPT_TLS) ||
(tlsOpts & PacketSocketFactory::OPT_TLS_INSECURE)) {
// Using TLS, wrap the socket in an SSL adapter.
SSLAdapter* ssl_adapter = SSLAdapter::Create(socket);
if (!ssl_adapter) {
return NULL;
}
if (tlsOpts & PacketSocketFactory::OPT_TLS_INSECURE) {
ssl_adapter->SetIgnoreBadCert(true);
}
ssl_adapter->SetAlpnProtocols(tcp_options.tls_alpn_protocols);
ssl_adapter->SetEllipticCurves(tcp_options.tls_elliptic_curves);
ssl_adapter->SetCertVerifier(tcp_options.tls_cert_verifier);
socket = ssl_adapter;
if (ssl_adapter->StartSSL(remote_address.hostname().c_str()) != 0) {
delete ssl_adapter;
return NULL;
}
} else if (tlsOpts & PacketSocketFactory::OPT_TLS_FAKE) {
// Using fake TLS, wrap the TCP socket in a pseudo-SSL socket.
socket = new AsyncSSLSocket(socket);
}
if (socket->Connect(remote_address) < 0) {
RTC_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 (tcp_options.opts & PacketSocketFactory::OPT_STUN) {
tcp_socket = new cricket::AsyncStunTCPSocket(socket);
} else {
tcp_socket = new AsyncTCPSocket(socket);
}
return tcp_socket;
}
AsyncResolverInterface* BasicPacketSocketFactory::CreateAsyncResolver() {
return new AsyncResolver();
}
std::unique_ptr<webrtc::AsyncDnsResolverInterface>
BasicPacketSocketFactory::CreateAsyncDnsResolver() {
return std::make_unique<webrtc::AsyncDnsResolver>();
}
int BasicPacketSocketFactory::BindSocket(Socket* 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(SocketAddress(local_address.ipaddr(), port));
}
}
return ret;
}
} // namespace rtc