| /* |
| * Copyright 2019 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. |
| */ |
| |
| #ifndef API_PACKET_SOCKET_FACTORY_H_ |
| #define API_PACKET_SOCKET_FACTORY_H_ |
| |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "api/async_dns_resolver.h" |
| #include "api/wrapping_async_dns_resolver.h" |
| #include "rtc_base/async_packet_socket.h" |
| #include "rtc_base/proxy_info.h" |
| #include "rtc_base/system/rtc_export.h" |
| |
| namespace rtc { |
| |
| class SSLCertificateVerifier; |
| class AsyncResolverInterface; |
| |
| struct PacketSocketTcpOptions { |
| PacketSocketTcpOptions() = default; |
| ~PacketSocketTcpOptions() = default; |
| |
| int opts = 0; |
| std::vector<std::string> tls_alpn_protocols; |
| std::vector<std::string> tls_elliptic_curves; |
| // An optional custom SSL certificate verifier that an API user can provide to |
| // inject their own certificate verification logic (not available to users |
| // outside of the WebRTC repo). |
| SSLCertificateVerifier* tls_cert_verifier = nullptr; |
| }; |
| |
| class RTC_EXPORT PacketSocketFactory { |
| public: |
| enum Options { |
| OPT_STUN = 0x04, |
| |
| // The TLS options below are mutually exclusive. |
| OPT_TLS = 0x02, // Real and secure TLS. |
| OPT_TLS_FAKE = 0x01, // Fake TLS with a dummy SSL handshake. |
| OPT_TLS_INSECURE = 0x08, // Insecure TLS without certificate validation. |
| |
| // Deprecated, use OPT_TLS_FAKE. |
| OPT_SSLTCP = OPT_TLS_FAKE, |
| }; |
| |
| PacketSocketFactory() = default; |
| virtual ~PacketSocketFactory() = default; |
| |
| virtual AsyncPacketSocket* CreateUdpSocket(const SocketAddress& address, |
| uint16_t min_port, |
| uint16_t max_port) = 0; |
| virtual AsyncListenSocket* CreateServerTcpSocket( |
| const SocketAddress& local_address, |
| uint16_t min_port, |
| uint16_t max_port, |
| int opts) = 0; |
| |
| virtual AsyncPacketSocket* CreateClientTcpSocket( |
| const SocketAddress& local_address, |
| const SocketAddress& remote_address, |
| const ProxyInfo& proxy_info, |
| const std::string& user_agent, |
| const PacketSocketTcpOptions& tcp_options) = 0; |
| |
| // The AsyncResolverInterface is deprecated; users are encouraged |
| // to switch to the AsyncDnsResolverInterface. |
| // TODO(bugs.webrtc.org/12598): Remove once all downstream users |
| // are converted. |
| #pragma clang diagnostic push |
| #pragma clang diagnostic ignored "-Wdeprecated-declarations" |
| [[deprecated]] virtual AsyncResolverInterface* CreateAsyncResolver() { |
| // Default implementation, so that downstream users can remove this |
| // immediately after changing to CreateAsyncDnsResolver |
| RTC_DCHECK_NOTREACHED(); |
| return nullptr; |
| } |
| |
| virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface> |
| CreateAsyncDnsResolver() { |
| // Default implementation, to aid in transition to AsyncDnsResolverInterface |
| return std::make_unique<webrtc::WrappingAsyncDnsResolver>( |
| CreateAsyncResolver()); |
| } |
| #pragma clang diagnostic pop |
| |
| private: |
| PacketSocketFactory(const PacketSocketFactory&) = delete; |
| PacketSocketFactory& operator=(const PacketSocketFactory&) = delete; |
| }; |
| |
| } // namespace rtc |
| |
| #endif // API_PACKET_SOCKET_FACTORY_H_ |