| /* | |
| * 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_SSL_STREAM_ADAPTER_H_ | |
| #define RTC_BASE_SSL_STREAM_ADAPTER_H_ | |
| #include <stddef.h> | |
| #include <stdint.h> | |
| #include <memory> | |
| #include <optional> | |
| #include <set> | |
| #include <string> | |
| #include <vector> | |
| #include "absl/functional/any_invocable.h" | |
| #include "absl/strings/string_view.h" | |
| #include "api/array_view.h" | |
| #include "api/field_trials_view.h" | |
| #include "rtc_base/buffer.h" | |
| #include "rtc_base/checks.h" | |
| #include "rtc_base/ssl_certificate.h" | |
| #include "rtc_base/ssl_identity.h" | |
| #include "rtc_base/stream.h" | |
| namespace webrtc { | |
| // Constants for SSL profile. | |
| constexpr int kTlsNullWithNullNull = 0; | |
| constexpr int kSslCipherSuiteMaxValue = 0xFFFF; | |
| // Constants for SRTP profiles. | |
| constexpr int kSrtpInvalidCryptoSuite = 0; | |
| constexpr int kSrtpAes128CmSha1_80 = 0x0001; | |
| constexpr int kSrtpAes128CmSha1_32 = 0x0002; | |
| constexpr int kSrtpAeadAes128Gcm = 0x0007; | |
| constexpr int kSrtpAeadAes256Gcm = 0x0008; | |
| constexpr int kSrtpCryptoSuiteMaxValue = 0xFFFF; | |
| // Constants for SSL signature algorithms. | |
| constexpr int kSslSignatureAlgorithmUnknown = 0; | |
| constexpr int kSslSignatureAlgorithmMaxValue = 0xFFFF; | |
| // Names of SRTP profiles listed above. | |
| // 128-bit AES with 80-bit SHA-1 HMAC. | |
| extern const char kCsAesCm128HmacSha1_80[]; | |
| // 128-bit AES with 32-bit SHA-1 HMAC. | |
| extern const char kCsAesCm128HmacSha1_32[]; | |
| // 128-bit AES GCM with 16 byte AEAD auth tag. | |
| extern const char kCsAeadAes128Gcm[]; | |
| // 256-bit AES GCM with 16 byte AEAD auth tag. | |
| extern const char kCsAeadAes256Gcm[]; | |
| // Given the DTLS-SRTP protection profile ID, as defined in | |
| // https://tools.ietf.org/html/rfc4568#section-6.2 , return the SRTP profile | |
| // name, as defined in https://tools.ietf.org/html/rfc5764#section-4.1.2. | |
| std::string SrtpCryptoSuiteToName(int crypto_suite); | |
| // Get key length and salt length for given crypto suite. Returns true for | |
| // valid suites, otherwise false. | |
| bool GetSrtpKeyAndSaltLengths(int crypto_suite, | |
| int* key_length, | |
| int* salt_length); | |
| // Returns true if the given crypto suite id uses a GCM cipher. | |
| bool IsGcmCryptoSuite(int crypto_suite); | |
| // SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS. | |
| // After SSL has been started, the stream will only open on successful | |
| // SSL verification of certificates, and the communication is | |
| // encrypted of course. | |
| // | |
| // This class was written with SSLAdapter as a starting point. It | |
| // offers a similar interface, with two differences: there is no | |
| // support for a restartable SSL connection, and this class has a | |
| // peer-to-peer mode. | |
| // | |
| // The SSL library requires initialization and cleanup. Static method | |
| // for doing this are in SSLAdapter. They should possibly be moved out | |
| // to a neutral class. | |
| enum SSLRole { SSL_CLIENT, SSL_SERVER }; | |
| enum SSLMode { SSL_MODE_TLS, SSL_MODE_DTLS }; | |
| // TODO bugs.webrtc.org/40644300 remove unused legacy constants. | |
| enum SSLProtocolVersion { | |
| SSL_PROTOCOL_NOT_GIVEN = -1, | |
| SSL_PROTOCOL_TLS_10 = 0, // Deprecated and no longer supported. | |
| SSL_PROTOCOL_TLS_11 = 1, // Deprecated and no longer supported. | |
| SSL_PROTOCOL_TLS_12 = 2, | |
| SSL_PROTOCOL_TLS_13 = 3, | |
| SSL_PROTOCOL_DTLS_10 = 1, // Deprecated and no longer supported. | |
| SSL_PROTOCOL_DTLS_12 = SSL_PROTOCOL_TLS_12, | |
| SSL_PROTOCOL_DTLS_13 = SSL_PROTOCOL_TLS_13, | |
| }; | |
| // Versions returned from BoringSSL. | |
| const uint16_t kDtls10VersionBytes = 0xfeff; | |
| const uint16_t kDtls12VersionBytes = 0xfefd; | |
| const uint16_t kDtls13VersionBytes = 0xfefc; | |
| enum class SSLPeerCertificateDigestError { | |
| NONE, | |
| UNKNOWN_ALGORITHM, | |
| INVALID_LENGTH, | |
| VERIFICATION_FAILED, | |
| }; | |
| // Errors for Read -- in the high range so no conflict with OpenSSL. | |
| enum { SSE_MSG_TRUNC = 0xff0001 }; | |
| // Used to send back UMA histogram value. Logged when Dtls handshake fails. | |
| enum class SSLHandshakeError { UNKNOWN, INCOMPATIBLE_CIPHERSUITE, MAX_VALUE }; | |
| class SSLStreamAdapter : public StreamInterface { | |
| public: | |
| // Instantiate an SSLStreamAdapter wrapping the given stream, | |
| // (using the selected implementation for the platform). | |
| // Caller is responsible for freeing the returned object. | |
| static std::unique_ptr<SSLStreamAdapter> Create( | |
| std::unique_ptr<StreamInterface> stream, | |
| absl::AnyInvocable<void(SSLHandshakeError)> handshake_error = nullptr, | |
| const FieldTrialsView* field_trials = nullptr); | |
| SSLStreamAdapter() = default; | |
| ~SSLStreamAdapter() override = default; | |
| // Specify our SSL identity: key and certificate. SSLStream takes ownership | |
| // of the SSLIdentity object and will free it when appropriate. Should be | |
| // called no more than once on a given SSLStream instance. | |
| virtual void SetIdentity(std::unique_ptr<SSLIdentity> identity) = 0; | |
| virtual SSLIdentity* GetIdentityForTesting() const = 0; | |
| // Call this to indicate that we are to play the server role (or client role, | |
| // if the default argument is replaced by SSL_CLIENT). | |
| // The default argument is for backward compatibility. | |
| // TODO(ekr@rtfm.com): rename this SetRole to reflect its new function | |
| virtual void SetServerRole(SSLRole role = SSL_SERVER) = 0; | |
| [[deprecated("Only DTLS is supported by the stream adapter")]] virtual void | |
| SetMode(SSLMode mode) = 0; | |
| // Set maximum supported protocol version. The highest version supported by | |
| // both ends will be used for the connection, i.e. if one party supports | |
| // DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used. | |
| // If requested version is not supported by underlying crypto library, the | |
| // next lower will be used. | |
| virtual void SetMaxProtocolVersion(SSLProtocolVersion version) = 0; | |
| // Set the initial retransmission timeout for DTLS messages. When the timeout | |
| // expires, the message gets retransmitted and the timeout is exponentially | |
| // increased. | |
| // This should only be called before StartSSL(). | |
| virtual void SetInitialRetransmissionTimeout(int timeout_ms) = 0; | |
| virtual void UpdateRetransmissionTimeout(int timeout_ms) {} | |
| // Set MTU to be used for next handshake flight. | |
| virtual void SetMTU(int mtu) = 0; | |
| // StartSSL starts negotiation with a peer, whose certificate is verified | |
| // using the certificate digest. Generally, SetIdentity() and possibly | |
| // SetServerRole() should have been called before this. | |
| // SetPeerCertificateDigest() must also be called. It may be called after | |
| // StartSSLWithPeer() but must be called before the underlying stream opens. | |
| // | |
| // Use of the stream prior to calling StartSSL will pass data in clear text. | |
| // Calling StartSSL causes SSL negotiation to begin as soon as possible: right | |
| // away if the underlying wrapped stream is already opened, or else as soon as | |
| // it opens. | |
| // | |
| // StartSSL returns a negative error code on failure. Returning 0 means | |
| // success so far, but negotiation is probably not complete and will continue | |
| // asynchronously. In that case, the exposed stream will open after | |
| // successful negotiation and verification, or an SE_CLOSE event will be | |
| // raised if negotiation fails. | |
| virtual int StartSSL() = 0; | |
| // Specify the digest of the certificate that our peer is expected to use. | |
| // Only this certificate will be accepted during SSL verification. The | |
| // certificate is assumed to have been obtained through some other secure | |
| // channel (such as the signaling channel). This must specify the terminal | |
| // certificate, not just a CA. SSLStream makes a copy of the digest value. | |
| // | |
| // Returns SSLPeerCertificateDigestError::NONE if successful. | |
| virtual SSLPeerCertificateDigestError SetPeerCertificateDigest( | |
| absl::string_view digest_alg, | |
| ArrayView<const uint8_t> digest_val) = 0; | |
| [[deprecated( | |
| "Use SetPeerCertificateDigest with ArrayView instead")]] virtual bool | |
| SetPeerCertificateDigest(absl::string_view digest_alg, | |
| const unsigned char* digest_val, | |
| size_t digest_len, | |
| SSLPeerCertificateDigestError* error = nullptr); | |
| // Retrieves the peer's certificate chain including leaf certificate, if a | |
| // connection has been established. | |
| virtual std::unique_ptr<SSLCertChain> GetPeerSSLCertChain() const = 0; | |
| // Retrieves the IANA registration id of the cipher suite used for the | |
| // connection (e.g. 0x2F for "TLS_RSA_WITH_AES_128_CBC_SHA"). | |
| virtual bool GetSslCipherSuite(int* cipher_suite) const = 0; | |
| // Returns the name of the cipher suite used for the DTLS transport, | |
| // as defined in the "Description" column of the IANA cipher suite registry. | |
| virtual std::optional<absl::string_view> GetTlsCipherSuiteName() const = 0; | |
| // Retrieves the enum value for SSL version. | |
| // Will return -1 until the version has been negotiated. | |
| [[deprecated("Use GetSslVersionBytes")]] virtual SSLProtocolVersion | |
| GetSslVersion() const = 0; | |
| // Retrieves the 2-byte version from the TLS protocol. | |
| // Will return false until the version has been negotiated. | |
| virtual bool GetSslVersionBytes(int* version) const = 0; | |
| // Key Exporter interface from RFC 5705 | |
| // The buffer must be preinitialized with a `size()` that will fit exactly | |
| // the keying material. | |
| [[deprecated("Use AppendSrtpKeyingMaterial")]] virtual bool | |
| ExportSrtpKeyingMaterial(ZeroOnFreeBuffer<uint8_t>& keying_material) { | |
| RTC_DCHECK_NOTREACHED() << "Use AppendSrtpKeyingMaterial"; | |
| return false; | |
| } | |
| // Extract the keys and append them to the buffer. The function will compute | |
| // the amount of keying material needed and append this many bytes. | |
| // The buffer size will grow by the number of bytes appended. | |
| virtual bool AppendSrtpKeyingMaterial( | |
| ZeroOnFreeBuffer<uint8_t>& keying_material) = 0; | |
| // Returns the signature algorithm or 0 if not applicable. | |
| virtual uint16_t GetPeerSignatureAlgorithm() const = 0; | |
| // DTLS-SRTP interface | |
| virtual bool SetDtlsSrtpCryptoSuites( | |
| const std::vector<int>& crypto_suites) = 0; | |
| virtual bool GetDtlsSrtpCryptoSuite(int* crypto_suite) const = 0; | |
| // Returns true if a TLS connection has been established. | |
| // The only difference between this and "GetState() == SE_OPEN" is that if | |
| // the peer certificate digest hasn't been verified, the state will still be | |
| // SS_OPENING but IsTlsConnected should return true. | |
| virtual bool IsTlsConnected() = 0; | |
| // Capabilities testing. | |
| // Used to have "DTLS supported", "DTLS-SRTP supported" etc. methods, but now | |
| // that's assumed. | |
| static bool IsBoringSsl(); | |
| // Returns true iff the supplied cipher is deemed to be strong. | |
| // TODO(torbjorng): Consider removing the KeyType argument. | |
| static bool IsAcceptableCipher(int cipher, KeyType key_type); | |
| static bool IsAcceptableCipher(absl::string_view cipher, KeyType key_type); | |
| static std::set<uint16_t> GetSupportedEphemeralKeyExchangeCipherGroups(); | |
| static std::optional<std::string> GetEphemeralKeyExchangeCipherGroupName( | |
| uint16_t group_id); | |
| static std::vector<uint16_t> GetDefaultEphemeralKeyExchangeCipherGroups( | |
| const FieldTrialsView* field_trials); | |
| //////////////////////////////////////////////////////////////////////////// | |
| // Testing only member functions | |
| //////////////////////////////////////////////////////////////////////////// | |
| // Use our timeutils.h source of timing in BoringSSL, allowing us to test | |
| // using a fake clock. | |
| static void EnableTimeCallbackForTesting(); | |
| // Return max DTLS SSLProtocolVersion supported by implementation. | |
| static SSLProtocolVersion GetMaxSupportedDTLSProtocolVersion(); | |
| // Deprecated. Do not use this API outside of testing. | |
| // Do not set this to false outside of testing. | |
| void SetClientAuthEnabledForTesting(bool enabled) { | |
| client_auth_enabled_ = enabled; | |
| } | |
| // Deprecated. Do not use this API outside of testing. | |
| // Returns true by default, else false if explicitly set to disable client | |
| // authentication. | |
| bool GetClientAuthEnabled() const { return client_auth_enabled_; } | |
| // Return number of times DTLS retransmission has been triggered. | |
| // Used for testing (and maybe put into stats?). | |
| virtual int GetRetransmissionCount() const = 0; | |
| // Set cipher group ids to use during DTLS handshake to establish ephemeral | |
| // key, see CryptoOptions::EphemeralKeyExchangeCipherGroups. | |
| virtual bool SetSslGroupIds(const std::vector<uint16_t>& group_ids) = 0; | |
| // Return the the ID of the group used by the adapters most recently | |
| // completed handshake, or 0 if not applicable (e.g. before the handshake). | |
| virtual uint16_t GetSslGroupId() const = 0; | |
| private: | |
| // If true (default), the client is required to provide a certificate during | |
| // handshake. If no certificate is given, handshake fails. This applies to | |
| // server mode only. | |
| bool client_auth_enabled_ = true; | |
| }; | |
| } // namespace webrtc | |
| #endif // RTC_BASE_SSL_STREAM_ADAPTER_H_ |