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+/*
+ * Copyright 2004 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 RTC_BASE_OPENSSL_STREAM_ADAPTER_H_
+#define RTC_BASE_OPENSSL_STREAM_ADAPTER_H_
+
+#include <openssl/ossl_typ.h>
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "rtc_base/buffer.h"
+#ifdef OPENSSL_IS_BORINGSSL
+#include "rtc_base/boringssl_identity.h"
+#else
+#include "rtc_base/openssl_identity.h"
+#endif
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "rtc_base/ssl_identity.h"
+#include "rtc_base/ssl_stream_adapter.h"
+#include "rtc_base/stream.h"
+#include "rtc_base/system/rtc_export.h"
+#include "rtc_base/task_utils/repeating_task.h"
+
+namespace rtc {
+
+// This class was written with OpenSSLAdapter (a socket adapter) as a
+// starting point. It has similar structure and functionality, but uses a
+// "peer-to-peer" mode, verifying the peer's certificate using a digest
+// sent over a secure signaling channel.
+//
+// Static methods to initialize and deinit the SSL library are in
+// OpenSSLAdapter. These should probably be moved out to a neutral class.
+//
+// In a few cases I have factored out some OpenSSLAdapter code into static
+// methods so it can be reused from this class. Eventually that code should
+// probably be moved to a common support class. Unfortunately there remain a
+// few duplicated sections of code. I have not done more restructuring because
+// I did not want to affect existing code that uses OpenSSLAdapter.
+//
+// This class does not support the SSL connection restart feature present in
+// OpenSSLAdapter. I am not entirely sure how the feature is useful and I am
+// not convinced that it works properly.
+//
+// This implementation is careful to disallow data exchange after an SSL error,
+// and it has an explicit SSL_CLOSED state. It should not be possible to send
+// any data in clear after one of the StartSSL methods has been called.
+
+// Look in sslstreamadapter.h for documentation of the methods.
+
+class SSLCertChain;
+
+///////////////////////////////////////////////////////////////////////////////
+
+// If `allow` has a value, its value determines if legacy TLS protocols are
+// allowed, overriding the default configuration.
+// If `allow` has no value, any previous override is removed and the default
+// configuration is restored.
+RTC_EXPORT void SetAllowLegacyTLSProtocols(const absl::optional<bool>& allow);
+
+class OpenSSLStreamAdapter final : public SSLStreamAdapter {
+ public:
+ explicit OpenSSLStreamAdapter(std::unique_ptr<StreamInterface> stream);
+ ~OpenSSLStreamAdapter() override;
+
+ void SetIdentity(std::unique_ptr<SSLIdentity> identity) override;
+ SSLIdentity* GetIdentityForTesting() const override;
+
+ // Default argument is for compatibility
+ void SetServerRole(SSLRole role = SSL_SERVER) override;
+ bool SetPeerCertificateDigest(
+ absl::string_view digest_alg,
+ const unsigned char* digest_val,
+ size_t digest_len,
+ SSLPeerCertificateDigestError* error = nullptr) override;
+
+ std::unique_ptr<SSLCertChain> GetPeerSSLCertChain() const override;
+
+ // Goes from state SSL_NONE to either SSL_CONNECTING or SSL_WAIT, depending
+ // on whether the underlying stream is already open or not.
+ int StartSSL() override;
+ void SetMode(SSLMode mode) override;
+ void SetMaxProtocolVersion(SSLProtocolVersion version) override;
+ void SetInitialRetransmissionTimeout(int timeout_ms) override;
+
+ StreamResult Read(void* data,
+ size_t data_len,
+ size_t* read,
+ int* error) override;
+ StreamResult Write(const void* data,
+ size_t data_len,
+ size_t* written,
+ int* error) override;
+ void Close() override;
+ StreamState GetState() const override;
+
+ // TODO(guoweis): Move this away from a static class method.
+ static std::string SslCipherSuiteToName(int crypto_suite);
+
+ bool GetSslCipherSuite(int* cipher) override;
+
+ SSLProtocolVersion GetSslVersion() const override;
+ bool GetSslVersionBytes(int* version) const override;
+ // Key Extractor interface
+ bool ExportKeyingMaterial(absl::string_view label,
+ const uint8_t* context,
+ size_t context_len,
+ bool use_context,
+ uint8_t* result,
+ size_t result_len) override;
+
+ // DTLS-SRTP interface
+ bool SetDtlsSrtpCryptoSuites(const std::vector<int>& crypto_suites) override;
+ bool GetDtlsSrtpCryptoSuite(int* crypto_suite) override;
+
+ bool IsTlsConnected() override;
+
+ // Capabilities interfaces.
+ static bool IsBoringSsl();
+
+ static bool IsAcceptableCipher(int cipher, KeyType key_type);
+ static bool IsAcceptableCipher(absl::string_view cipher, KeyType key_type);
+
+ // Use our timeutils.h source of timing in BoringSSL, allowing us to test
+ // using a fake clock.
+ static void EnableTimeCallbackForTesting();
+
+ private:
+ enum SSLState {
+ // Before calling one of the StartSSL methods, data flows
+ // in clear text.
+ SSL_NONE,
+ SSL_WAIT, // waiting for the stream to open to start SSL negotiation
+ SSL_CONNECTING, // SSL negotiation in progress
+ SSL_CONNECTED, // SSL stream successfully established
+ SSL_ERROR, // some SSL error occurred, stream is closed
+ SSL_CLOSED // Clean close
+ };
+
+ void OnEvent(StreamInterface* stream, int events, int err);
+
+ void PostEvent(int events, int err);
+ void SetTimeout(int delay_ms);
+
+ // The following three methods return 0 on success and a negative
+ // error code on failure. The error code may be from OpenSSL or -1
+ // on some other error cases, so it can't really be interpreted
+ // unfortunately.
+
+ // Prepare SSL library, state is SSL_CONNECTING.
+ int BeginSSL();
+ // Perform SSL negotiation steps.
+ int ContinueSSL();
+
+ // Error handler helper. signal is given as true for errors in
+ // asynchronous contexts (when an error method was not returned
+ // through some other method), and in that case an SE_CLOSE event is
+ // raised on the stream with the specified error.
+ // A 0 error means a graceful close, otherwise there is not really enough
+ // context to interpret the error code.
+ // `alert` indicates an alert description (one of the SSL_AD constants) to
+ // send to the remote endpoint when closing the association. If 0, a normal
+ // shutdown will be performed.
+ void Error(absl::string_view context, int err, uint8_t alert, bool signal);
+ void Cleanup(uint8_t alert);
+
+ // Flush the input buffers by reading left bytes (for DTLS)
+ void FlushInput(unsigned int left);
+
+ // SSL library configuration
+ SSL_CTX* SetupSSLContext();
+ // Verify the peer certificate matches the signaled digest.
+ bool VerifyPeerCertificate();
+
+#ifdef OPENSSL_IS_BORINGSSL
+ // SSL certificate verification callback. See SSL_CTX_set_custom_verify.
+ static enum ssl_verify_result_t SSLVerifyCallback(SSL* ssl,
+ uint8_t* out_alert);
+#else
+ // SSL certificate verification callback. See
+ // SSL_CTX_set_cert_verify_callback.
+ static int SSLVerifyCallback(X509_STORE_CTX* store, void* arg);
+#endif
+
+ bool WaitingToVerifyPeerCertificate() const {
+ return GetClientAuthEnabled() && !peer_certificate_verified_;
+ }
+
+ bool HasPeerCertificateDigest() const {
+ return !peer_certificate_digest_algorithm_.empty() &&
+ !peer_certificate_digest_value_.empty();
+ }
+
+ const std::unique_ptr<StreamInterface> stream_;
+
+ rtc::Thread* const owner_;
+ webrtc::ScopedTaskSafety task_safety_;
+ webrtc::RepeatingTaskHandle timeout_task_;
+
+ SSLState state_;
+ SSLRole role_;
+ int ssl_error_code_; // valid when state_ == SSL_ERROR or SSL_CLOSED
+ // Whether the SSL negotiation is blocked on needing to read or
+ // write to the wrapped stream.
+ bool ssl_read_needs_write_;
+ bool ssl_write_needs_read_;
+
+ SSL* ssl_;
+ SSL_CTX* ssl_ctx_;
+
+ // Our key and certificate.
+#ifdef OPENSSL_IS_BORINGSSL
+ std::unique_ptr<BoringSSLIdentity> identity_;
+#else
+ std::unique_ptr<OpenSSLIdentity> identity_;
+#endif
+ // The certificate chain that the peer presented. Initially null, until the
+ // connection is established.
+ std::unique_ptr<SSLCertChain> peer_cert_chain_;
+ bool peer_certificate_verified_ = false;
+ // The digest of the certificate that the peer must present.
+ Buffer peer_certificate_digest_value_;
+ std::string peer_certificate_digest_algorithm_;
+
+ // The DtlsSrtp ciphers
+ std::string srtp_ciphers_;
+
+ // Do DTLS or not
+ SSLMode ssl_mode_;
+
+ // Max. allowed protocol version
+ SSLProtocolVersion ssl_max_version_;
+
+ // A 50-ms initial timeout ensures rapid setup on fast connections, but may
+ // be too aggressive for low bandwidth links.
+ int dtls_handshake_timeout_ms_ = 50;
+
+ // TODO(https://bugs.webrtc.org/10261): Completely remove this option in M84.
+ const bool support_legacy_tls_protocols_flag_;
+};
+
+/////////////////////////////////////////////////////////////////////////////
+
+} // namespace rtc
+
+#endif // RTC_BASE_OPENSSL_STREAM_ADAPTER_H_