|  | /* | 
|  | *  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 <string> | 
|  | #include <vector> | 
|  |  | 
|  | #include "absl/memory/memory.h" | 
|  | #include "absl/strings/string_view.h" | 
|  | #include "rtc_base/ssl_certificate.h" | 
|  | #include "rtc_base/ssl_identity.h" | 
|  | #include "rtc_base/stream.h" | 
|  | #include "rtc_base/third_party/sigslot/sigslot.h" | 
|  |  | 
|  | namespace rtc { | 
|  |  | 
|  | // 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; | 
|  |  | 
|  | // 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); | 
|  |  | 
|  | // The reverse of above conversion. | 
|  | int SrtpCryptoSuiteFromName(absl::string_view 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); | 
|  |  | 
|  | // Returns true if the given crypto suite name uses a GCM cipher. | 
|  | bool IsGcmCryptoSuiteName(absl::string_view 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 }; | 
|  |  | 
|  | // Note: TLS_10, TLS_11, and DTLS_10 will all be ignored, and only DTLS1_2 will | 
|  | // be accepted unless the trial flag WebRTC-LegacyTlsProtocols/Enabled/ is | 
|  | // passed in or an explicit override is used. Support for the legacy protocol | 
|  | // versions will be completely removed in the future. | 
|  | // See https://bugs.webrtc.org/10261. | 
|  | enum SSLProtocolVersion { | 
|  | SSL_PROTOCOL_NOT_GIVEN = -1, | 
|  | SSL_PROTOCOL_TLS_10 = 0, | 
|  | SSL_PROTOCOL_TLS_11, | 
|  | SSL_PROTOCOL_TLS_12, | 
|  | SSL_PROTOCOL_DTLS_10 = SSL_PROTOCOL_TLS_11, | 
|  | SSL_PROTOCOL_DTLS_12 = SSL_PROTOCOL_TLS_12, | 
|  | }; | 
|  | 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 sigslot::has_slots<> { | 
|  | 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); | 
|  |  | 
|  | 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; | 
|  |  | 
|  | // Do DTLS or TLS. | 
|  | 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; | 
|  |  | 
|  | // 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 true if successful. | 
|  | // `error` is optional and provides more information about the failure. | 
|  | virtual bool SetPeerCertificateDigest( | 
|  | absl::string_view digest_alg, | 
|  | const unsigned char* digest_val, | 
|  | size_t digest_len, | 
|  | SSLPeerCertificateDigestError* error = nullptr) = 0; | 
|  |  | 
|  | // 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); | 
|  |  | 
|  | // Retrieves the enum value for SSL version. | 
|  | // Will return -1 until the version has been negotiated. | 
|  | 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 | 
|  | // Arguments are: | 
|  | // label               -- the exporter label. | 
|  | //                        part of the RFC defining each exporter | 
|  | //                        usage (IN) | 
|  | // context/context_len -- a context to bind to for this connection; | 
|  | //                        optional, can be null, 0 (IN) | 
|  | // use_context         -- whether to use the context value | 
|  | //                        (needed to distinguish no context from | 
|  | //                        zero-length ones). | 
|  | // result              -- where to put the computed value | 
|  | // result_len          -- the length of the computed value | 
|  | virtual bool ExportKeyingMaterial(absl::string_view label, | 
|  | const uint8_t* context, | 
|  | size_t context_len, | 
|  | bool use_context, | 
|  | uint8_t* result, | 
|  | size_t result_len); | 
|  |  | 
|  | // DTLS-SRTP interface | 
|  | virtual bool SetDtlsSrtpCryptoSuites(const std::vector<int>& crypto_suites); | 
|  | virtual bool GetDtlsSrtpCryptoSuite(int* crypto_suite); | 
|  |  | 
|  | // 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); | 
|  |  | 
|  | // TODO(guoweis): Move this away from a static class method. Currently this is | 
|  | // introduced such that any caller could depend on sslstreamadapter.h without | 
|  | // depending on specific SSL implementation. | 
|  | static std::string SslCipherSuiteToName(int cipher_suite); | 
|  |  | 
|  | //////////////////////////////////////////////////////////////////////////// | 
|  | // Testing only member functions | 
|  | //////////////////////////////////////////////////////////////////////////// | 
|  |  | 
|  | // Use our timeutils.h source of timing in BoringSSL, allowing us to test | 
|  | // using a fake clock. | 
|  | static void EnableTimeCallbackForTesting(); | 
|  |  | 
|  | // 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_; } | 
|  |  | 
|  | sigslot::signal1<SSLHandshakeError> SignalSSLHandshakeError; | 
|  |  | 
|  | 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 rtc | 
|  |  | 
|  | #endif  // RTC_BASE_SSL_STREAM_ADAPTER_H_ |