blob: 4cd2b513663675a584fad0c0db3ff14b9fe019e0 [file] [log] [blame]
/*
* Copyright (c) 2020 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 "test/network/emulated_turn_server.h"
#include <string>
#include <utility>
#include "api/packet_socket_factory.h"
#include "rtc_base/strings/string_builder.h"
namespace {
static const char kTestRealm[] = "example.org";
static const char kTestSoftware[] = "TestTurnServer";
// A wrapper class for copying data between an AsyncPacketSocket and a
// EmulatedEndpoint. This is used by the cricket::TurnServer when
// sending data back into the emulated network.
class AsyncPacketSocketWrapper : public rtc::AsyncPacketSocket {
public:
AsyncPacketSocketWrapper(webrtc::test::EmulatedTURNServer* turn_server,
webrtc::EmulatedEndpoint* endpoint,
uint16_t port)
: turn_server_(turn_server),
endpoint_(endpoint),
local_address_(
rtc::SocketAddress(endpoint_->GetPeerLocalAddress(), port)) {}
~AsyncPacketSocketWrapper() { turn_server_->Unbind(local_address_); }
rtc::SocketAddress GetLocalAddress() const override { return local_address_; }
rtc::SocketAddress GetRemoteAddress() const override {
return rtc::SocketAddress();
}
int Send(const void* pv,
size_t cb,
const rtc::PacketOptions& options) override {
RTC_CHECK(false) << "TCP not implemented";
return -1;
}
int SendTo(const void* pv,
size_t cb,
const rtc::SocketAddress& addr,
const rtc::PacketOptions& options) override {
// Copy from rtc::AsyncPacketSocket to EmulatedEndpoint.
rtc::CopyOnWriteBuffer buf(reinterpret_cast<const char*>(pv), cb);
endpoint_->SendPacket(local_address_, addr, buf);
return cb;
}
int Close() override { return 0; }
rtc::AsyncPacketSocket::State GetState() const override {
return rtc::AsyncPacketSocket::STATE_BOUND;
}
int GetOption(rtc::Socket::Option opt, int* value) override { return 0; }
int SetOption(rtc::Socket::Option opt, int value) override { return 0; }
int GetError() const override { return 0; }
void SetError(int error) override {}
private:
webrtc::test::EmulatedTURNServer* const turn_server_;
webrtc::EmulatedEndpoint* const endpoint_;
const rtc::SocketAddress local_address_;
};
// A wrapper class for cricket::TurnServer to allocate sockets.
class PacketSocketFactoryWrapper : public rtc::PacketSocketFactory {
public:
explicit PacketSocketFactoryWrapper(
webrtc::test::EmulatedTURNServer* turn_server)
: turn_server_(turn_server) {}
~PacketSocketFactoryWrapper() override {}
// This method is called from TurnServer when making a TURN ALLOCATION.
// It will create a socket on the `peer_` endpoint.
rtc::AsyncPacketSocket* CreateUdpSocket(const rtc::SocketAddress& address,
uint16_t min_port,
uint16_t max_port) override {
return turn_server_->CreatePeerSocket();
}
rtc::AsyncListenSocket* CreateServerTcpSocket(
const rtc::SocketAddress& local_address,
uint16_t min_port,
uint16_t max_port,
int opts) override {
return nullptr;
}
rtc::AsyncPacketSocket* CreateClientTcpSocket(
const rtc::SocketAddress& local_address,
const rtc::SocketAddress& remote_address,
const rtc::ProxyInfo& proxy_info,
const std::string& user_agent,
const rtc::PacketSocketTcpOptions& tcp_options) override {
return nullptr;
}
rtc::AsyncResolverInterface* CreateAsyncResolver() override {
return nullptr;
}
private:
webrtc::test::EmulatedTURNServer* turn_server_;
};
} // namespace
namespace webrtc {
namespace test {
EmulatedTURNServer::EmulatedTURNServer(std::unique_ptr<rtc::Thread> thread,
EmulatedEndpoint* client,
EmulatedEndpoint* peer)
: thread_(std::move(thread)), client_(client), peer_(peer) {
ice_config_.username = "keso";
ice_config_.password = "keso";
thread_->Invoke<void>(RTC_FROM_HERE, [=]() {
RTC_DCHECK_RUN_ON(thread_.get());
turn_server_ = std::make_unique<cricket::TurnServer>(thread_.get());
turn_server_->set_realm(kTestRealm);
turn_server_->set_realm(kTestSoftware);
turn_server_->set_auth_hook(this);
auto client_socket = Wrap(client_);
turn_server_->AddInternalSocket(client_socket, cricket::PROTO_UDP);
turn_server_->SetExternalSocketFactory(new PacketSocketFactoryWrapper(this),
rtc::SocketAddress());
client_address_ = client_socket->GetLocalAddress();
char buf[256];
rtc::SimpleStringBuilder str(buf);
str.AppendFormat("turn:%s?transport=udp",
client_address_.ToString().c_str());
ice_config_.url = str.str();
});
}
void EmulatedTURNServer::Stop() {
thread_->Invoke<void>(RTC_FROM_HERE, [=]() {
RTC_DCHECK_RUN_ON(thread_.get());
sockets_.clear();
});
}
EmulatedTURNServer::~EmulatedTURNServer() {
thread_->Invoke<void>(RTC_FROM_HERE, [=]() {
RTC_DCHECK_RUN_ON(thread_.get());
turn_server_.reset(nullptr);
});
}
rtc::AsyncPacketSocket* EmulatedTURNServer::Wrap(EmulatedEndpoint* endpoint) {
RTC_DCHECK_RUN_ON(thread_.get());
auto port = endpoint->BindReceiver(0, this).value();
auto socket = new AsyncPacketSocketWrapper(this, endpoint, port);
sockets_[rtc::SocketAddress(endpoint->GetPeerLocalAddress(), port)] = socket;
return socket;
}
void EmulatedTURNServer::OnPacketReceived(webrtc::EmulatedIpPacket packet) {
// Copy from EmulatedEndpoint to rtc::AsyncPacketSocket.
thread_->PostTask(RTC_FROM_HERE, [this, packet(std::move(packet))]() {
RTC_DCHECK_RUN_ON(thread_.get());
auto it = sockets_.find(packet.to);
if (it != sockets_.end()) {
it->second->SignalReadPacket(
it->second, reinterpret_cast<const char*>(packet.cdata()),
packet.size(), packet.from, packet.arrival_time.ms());
}
});
}
void EmulatedTURNServer::Unbind(rtc::SocketAddress address) {
RTC_DCHECK_RUN_ON(thread_.get());
if (GetClientEndpoint()->GetPeerLocalAddress() == address.ipaddr()) {
GetClientEndpoint()->UnbindReceiver(address.port());
} else {
GetPeerEndpoint()->UnbindReceiver(address.port());
}
sockets_.erase(address);
}
} // namespace test
} // namespace webrtc