1 /* 2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef RTC_BASE_SSL_STREAM_ADAPTER_H_ 12 #define RTC_BASE_SSL_STREAM_ADAPTER_H_ 13 14 #include <stddef.h> 15 #include <stdint.h> 16 #include <memory> 17 #include <string> 18 #include <vector> 19 20 #include "absl/memory/memory.h" 21 #include "rtc_base/deprecation.h" 22 #include "rtc_base/ssl_certificate.h" 23 #include "rtc_base/ssl_identity.h" 24 #include "rtc_base/stream.h" 25 #include "rtc_base/third_party/sigslot/sigslot.h" 26 27 namespace rtc { 28 29 // Constants for SSL profile. 30 const int TLS_NULL_WITH_NULL_NULL = 0; 31 const int SSL_CIPHER_SUITE_MAX_VALUE = 0xFFFF; 32 33 // Constants for SRTP profiles. 34 const int SRTP_INVALID_CRYPTO_SUITE = 0; 35 #ifndef SRTP_AES128_CM_SHA1_80 36 const int SRTP_AES128_CM_SHA1_80 = 0x0001; 37 #endif 38 #ifndef SRTP_AES128_CM_SHA1_32 39 const int SRTP_AES128_CM_SHA1_32 = 0x0002; 40 #endif 41 #ifndef SRTP_AEAD_AES_128_GCM 42 const int SRTP_AEAD_AES_128_GCM = 0x0007; 43 #endif 44 #ifndef SRTP_AEAD_AES_256_GCM 45 const int SRTP_AEAD_AES_256_GCM = 0x0008; 46 #endif 47 const int SRTP_CRYPTO_SUITE_MAX_VALUE = 0xFFFF; 48 49 // Names of SRTP profiles listed above. 50 // 128-bit AES with 80-bit SHA-1 HMAC. 51 extern const char CS_AES_CM_128_HMAC_SHA1_80[]; 52 // 128-bit AES with 32-bit SHA-1 HMAC. 53 extern const char CS_AES_CM_128_HMAC_SHA1_32[]; 54 // 128-bit AES GCM with 16 byte AEAD auth tag. 55 extern const char CS_AEAD_AES_128_GCM[]; 56 // 256-bit AES GCM with 16 byte AEAD auth tag. 57 extern const char CS_AEAD_AES_256_GCM[]; 58 59 // Given the DTLS-SRTP protection profile ID, as defined in 60 // https://tools.ietf.org/html/rfc4568#section-6.2 , return the SRTP profile 61 // name, as defined in https://tools.ietf.org/html/rfc5764#section-4.1.2. 62 std::string SrtpCryptoSuiteToName(int crypto_suite); 63 64 // The reverse of above conversion. 65 int SrtpCryptoSuiteFromName(const std::string& crypto_suite); 66 67 // Get key length and salt length for given crypto suite. Returns true for 68 // valid suites, otherwise false. 69 bool GetSrtpKeyAndSaltLengths(int crypto_suite, 70 int* key_length, 71 int* salt_length); 72 73 // Returns true if the given crypto suite id uses a GCM cipher. 74 bool IsGcmCryptoSuite(int crypto_suite); 75 76 // Returns true if the given crypto suite name uses a GCM cipher. 77 bool IsGcmCryptoSuiteName(const std::string& crypto_suite); 78 79 // SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS. 80 // After SSL has been started, the stream will only open on successful 81 // SSL verification of certificates, and the communication is 82 // encrypted of course. 83 // 84 // This class was written with SSLAdapter as a starting point. It 85 // offers a similar interface, with two differences: there is no 86 // support for a restartable SSL connection, and this class has a 87 // peer-to-peer mode. 88 // 89 // The SSL library requires initialization and cleanup. Static method 90 // for doing this are in SSLAdapter. They should possibly be moved out 91 // to a neutral class. 92 93 enum SSLRole { SSL_CLIENT, SSL_SERVER }; 94 enum SSLMode { SSL_MODE_TLS, SSL_MODE_DTLS }; 95 96 // Note: TLS_10, TLS_11, and DTLS_10 will all be ignored, and only 97 // DTLS1_2 will be accepted, if the trial flag 98 // WebRTC-LegacyTlsProtocols/Disabled/ is passed in. Support for these 99 // protocol versions will be completely removed in M84 or later. 100 // TODO(https://bugs.webrtc.org/10261). 101 enum SSLProtocolVersion { 102 SSL_PROTOCOL_NOT_GIVEN = -1, 103 SSL_PROTOCOL_TLS_10 = 0, 104 SSL_PROTOCOL_TLS_11, 105 SSL_PROTOCOL_TLS_12, 106 SSL_PROTOCOL_DTLS_10 = SSL_PROTOCOL_TLS_11, 107 SSL_PROTOCOL_DTLS_12 = SSL_PROTOCOL_TLS_12, 108 }; 109 enum class SSLPeerCertificateDigestError { 110 NONE, 111 UNKNOWN_ALGORITHM, 112 INVALID_LENGTH, 113 VERIFICATION_FAILED, 114 }; 115 116 // Errors for Read -- in the high range so no conflict with OpenSSL. 117 enum { SSE_MSG_TRUNC = 0xff0001 }; 118 119 // Used to send back UMA histogram value. Logged when Dtls handshake fails. 120 enum class SSLHandshakeError { UNKNOWN, INCOMPATIBLE_CIPHERSUITE, MAX_VALUE }; 121 122 class SSLStreamAdapter : public StreamAdapterInterface { 123 public: 124 // Instantiate an SSLStreamAdapter wrapping the given stream, 125 // (using the selected implementation for the platform). 126 // Caller is responsible for freeing the returned object. 127 static std::unique_ptr<SSLStreamAdapter> Create( 128 std::unique_ptr<StreamInterface> stream); 129 130 explicit SSLStreamAdapter(std::unique_ptr<StreamInterface> stream); 131 ~SSLStreamAdapter() override; 132 133 // Specify our SSL identity: key and certificate. SSLStream takes ownership 134 // of the SSLIdentity object and will free it when appropriate. Should be 135 // called no more than once on a given SSLStream instance. 136 virtual void SetIdentity(std::unique_ptr<SSLIdentity> identity) = 0; 137 virtual SSLIdentity* GetIdentityForTesting() const = 0; 138 139 // Call this to indicate that we are to play the server role (or client role, 140 // if the default argument is replaced by SSL_CLIENT). 141 // The default argument is for backward compatibility. 142 // TODO(ekr@rtfm.com): rename this SetRole to reflect its new function 143 virtual void SetServerRole(SSLRole role = SSL_SERVER) = 0; 144 145 // Do DTLS or TLS. 146 virtual void SetMode(SSLMode mode) = 0; 147 148 // Set maximum supported protocol version. The highest version supported by 149 // both ends will be used for the connection, i.e. if one party supports 150 // DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used. 151 // If requested version is not supported by underlying crypto library, the 152 // next lower will be used. 153 virtual void SetMaxProtocolVersion(SSLProtocolVersion version) = 0; 154 155 // Set the initial retransmission timeout for DTLS messages. When the timeout 156 // expires, the message gets retransmitted and the timeout is exponentially 157 // increased. 158 // This should only be called before StartSSL(). 159 virtual void SetInitialRetransmissionTimeout(int timeout_ms) = 0; 160 161 // StartSSL starts negotiation with a peer, whose certificate is verified 162 // using the certificate digest. Generally, SetIdentity() and possibly 163 // SetServerRole() should have been called before this. 164 // SetPeerCertificateDigest() must also be called. It may be called after 165 // StartSSLWithPeer() but must be called before the underlying stream opens. 166 // 167 // Use of the stream prior to calling StartSSL will pass data in clear text. 168 // Calling StartSSL causes SSL negotiation to begin as soon as possible: right 169 // away if the underlying wrapped stream is already opened, or else as soon as 170 // it opens. 171 // 172 // StartSSL returns a negative error code on failure. Returning 0 means 173 // success so far, but negotiation is probably not complete and will continue 174 // asynchronously. In that case, the exposed stream will open after 175 // successful negotiation and verification, or an SE_CLOSE event will be 176 // raised if negotiation fails. 177 virtual int StartSSL() = 0; 178 179 // Specify the digest of the certificate that our peer is expected to use. 180 // Only this certificate will be accepted during SSL verification. The 181 // certificate is assumed to have been obtained through some other secure 182 // channel (such as the signaling channel). This must specify the terminal 183 // certificate, not just a CA. SSLStream makes a copy of the digest value. 184 // 185 // Returns true if successful. 186 // |error| is optional and provides more information about the failure. 187 virtual bool SetPeerCertificateDigest( 188 const std::string& digest_alg, 189 const unsigned char* digest_val, 190 size_t digest_len, 191 SSLPeerCertificateDigestError* error = nullptr) = 0; 192 193 // Retrieves the peer's certificate chain including leaf certificate, if a 194 // connection has been established. 195 virtual std::unique_ptr<SSLCertChain> GetPeerSSLCertChain() const = 0; 196 197 // Retrieves the IANA registration id of the cipher suite used for the 198 // connection (e.g. 0x2F for "TLS_RSA_WITH_AES_128_CBC_SHA"). 199 virtual bool GetSslCipherSuite(int* cipher_suite); 200 201 // Retrieves the enum value for SSL version. 202 // Will return -1 until the version has been negotiated. 203 virtual SSLProtocolVersion GetSslVersion() const = 0; 204 // Retrieves the 2-byte version from the TLS protocol. 205 // Will return false until the version has been negotiated. 206 virtual bool GetSslVersionBytes(int* version) const = 0; 207 208 // Key Exporter interface from RFC 5705 209 // Arguments are: 210 // label -- the exporter label. 211 // part of the RFC defining each exporter 212 // usage (IN) 213 // context/context_len -- a context to bind to for this connection; 214 // optional, can be null, 0 (IN) 215 // use_context -- whether to use the context value 216 // (needed to distinguish no context from 217 // zero-length ones). 218 // result -- where to put the computed value 219 // result_len -- the length of the computed value 220 virtual bool ExportKeyingMaterial(const std::string& label, 221 const uint8_t* context, 222 size_t context_len, 223 bool use_context, 224 uint8_t* result, 225 size_t result_len); 226 227 // DTLS-SRTP interface 228 virtual bool SetDtlsSrtpCryptoSuites(const std::vector<int>& crypto_suites); 229 virtual bool GetDtlsSrtpCryptoSuite(int* crypto_suite); 230 231 // Returns true if a TLS connection has been established. 232 // The only difference between this and "GetState() == SE_OPEN" is that if 233 // the peer certificate digest hasn't been verified, the state will still be 234 // SS_OPENING but IsTlsConnected should return true. 235 virtual bool IsTlsConnected() = 0; 236 237 // Capabilities testing. 238 // Used to have "DTLS supported", "DTLS-SRTP supported" etc. methods, but now 239 // that's assumed. 240 static bool IsBoringSsl(); 241 242 // Returns true iff the supplied cipher is deemed to be strong. 243 // TODO(torbjorng): Consider removing the KeyType argument. 244 static bool IsAcceptableCipher(int cipher, KeyType key_type); 245 static bool IsAcceptableCipher(const std::string& cipher, KeyType key_type); 246 247 // TODO(guoweis): Move this away from a static class method. Currently this is 248 // introduced such that any caller could depend on sslstreamadapter.h without 249 // depending on specific SSL implementation. 250 static std::string SslCipherSuiteToName(int cipher_suite); 251 252 //////////////////////////////////////////////////////////////////////////// 253 // Testing only member functions 254 //////////////////////////////////////////////////////////////////////////// 255 256 // Use our timeutils.h source of timing in BoringSSL, allowing us to test 257 // using a fake clock. 258 static void EnableTimeCallbackForTesting(); 259 260 // Deprecated. Do not use this API outside of testing. 261 // Do not set this to false outside of testing. SetClientAuthEnabledForTesting(bool enabled)262 void SetClientAuthEnabledForTesting(bool enabled) { 263 client_auth_enabled_ = enabled; 264 } 265 266 // Deprecated. Do not use this API outside of testing. 267 // Returns true by default, else false if explicitly set to disable client 268 // authentication. GetClientAuthEnabled()269 bool GetClientAuthEnabled() const { return client_auth_enabled_; } 270 271 sigslot::signal1<SSLHandshakeError> SignalSSLHandshakeError; 272 273 private: 274 // If true (default), the client is required to provide a certificate during 275 // handshake. If no certificate is given, handshake fails. This applies to 276 // server mode only. 277 bool client_auth_enabled_ = true; 278 }; 279 280 } // namespace rtc 281 282 #endif // RTC_BASE_SSL_STREAM_ADAPTER_H_ 283