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| 1 /* | 1 /* |
| 2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. | 2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| 11 #ifndef WEBRTC_BASE_SSLSTREAMADAPTER_H_ | 11 #ifndef WEBRTC_BASE_SSLSTREAMADAPTER_H_ |
| 12 #define WEBRTC_BASE_SSLSTREAMADAPTER_H_ | 12 #define WEBRTC_BASE_SSLSTREAMADAPTER_H_ |
| 13 | 13 |
| 14 #include <string> | 14 #include <string> |
| 15 #include <vector> | 15 #include <vector> |
| 16 | 16 |
| 17 #include "webrtc/base/stream.h" | 17 #include "webrtc/base/stream.h" |
| 18 #include "webrtc/base/sslidentity.h" | 18 #include "webrtc/base/sslidentity.h" |
| 19 | 19 |
| 20 namespace rtc { | 20 namespace rtc { |
| 21 | 21 |
| 22 // Constants for SRTP profiles. |
| 23 const uint16_t SRTP_AES128_CM_SHA1_80 = 0x0001; |
| 24 const uint16_t SRTP_AES128_CM_SHA1_32 = 0x0002; |
| 25 |
| 26 // Cipher suite to use for SRTP. Typically a 80-bit HMAC will be used, except |
| 27 // in applications (voice) where the additional bandwidth may be significant. |
| 28 // A 80-bit HMAC is always used for SRTCP. |
| 29 // 128-bit AES with 80-bit SHA-1 HMAC. |
| 30 extern const char CS_AES_CM_128_HMAC_SHA1_80[]; |
| 31 // 128-bit AES with 32-bit SHA-1 HMAC. |
| 32 extern const char CS_AES_CM_128_HMAC_SHA1_32[]; |
| 33 |
| 34 // Returns the DTLS-SRTP protection profile ID, as defined in |
| 35 // https://tools.ietf.org/html/rfc5764#section-4.1.2, for the given SRTP |
| 36 // Crypto-suite, as defined in https://tools.ietf.org/html/rfc4568#section-6.2 |
| 37 uint16_t GetSrtpCryptoSuiteFromName(const std::string& cipher_rfc_name); |
| 38 |
| 22 // SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS. | 39 // SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS. |
| 23 // After SSL has been started, the stream will only open on successful | 40 // After SSL has been started, the stream will only open on successful |
| 24 // SSL verification of certificates, and the communication is | 41 // SSL verification of certificates, and the communication is |
| 25 // encrypted of course. | 42 // encrypted of course. |
| 26 // | 43 // |
| 27 // This class was written with SSLAdapter as a starting point. It | 44 // This class was written with SSLAdapter as a starting point. It |
| 28 // offers a similar interface, with two differences: there is no | 45 // offers a similar interface, with two differences: there is no |
| 29 // support for a restartable SSL connection, and this class has a | 46 // support for a restartable SSL connection, and this class has a |
| 30 // peer-to-peer mode. | 47 // peer-to-peer mode. |
| 31 // | 48 // |
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| 126 // SSLStream makes a copy of the digest value. | 143 // SSLStream makes a copy of the digest value. |
| 127 virtual bool SetPeerCertificateDigest(const std::string& digest_alg, | 144 virtual bool SetPeerCertificateDigest(const std::string& digest_alg, |
| 128 const unsigned char* digest_val, | 145 const unsigned char* digest_val, |
| 129 size_t digest_len) = 0; | 146 size_t digest_len) = 0; |
| 130 | 147 |
| 131 // Retrieves the peer's X.509 certificate, if a connection has been | 148 // Retrieves the peer's X.509 certificate, if a connection has been |
| 132 // established. It returns the transmitted over SSL, including the entire | 149 // established. It returns the transmitted over SSL, including the entire |
| 133 // chain. The returned certificate is owned by the caller. | 150 // chain. The returned certificate is owned by the caller. |
| 134 virtual bool GetPeerCertificate(SSLCertificate** cert) const = 0; | 151 virtual bool GetPeerCertificate(SSLCertificate** cert) const = 0; |
| 135 | 152 |
| 136 // Retrieves the name of the cipher suite used for the connection | 153 // Retrieves the IANA registration id of the cipher suite used for the |
| 137 // (e.g. "TLS_RSA_WITH_AES_128_CBC_SHA"). | 154 // connection (e.g. 0x2F for "TLS_RSA_WITH_AES_128_CBC_SHA"). |
| 138 virtual bool GetSslCipher(std::string* cipher); | 155 virtual bool GetSslCipherSuite(uint16_t* cipher); |
| 139 | 156 |
| 140 // Key Exporter interface from RFC 5705 | 157 // Key Exporter interface from RFC 5705 |
| 141 // Arguments are: | 158 // Arguments are: |
| 142 // label -- the exporter label. | 159 // label -- the exporter label. |
| 143 // part of the RFC defining each exporter | 160 // part of the RFC defining each exporter |
| 144 // usage (IN) | 161 // usage (IN) |
| 145 // context/context_len -- a context to bind to for this connection; | 162 // context/context_len -- a context to bind to for this connection; |
| 146 // optional, can be NULL, 0 (IN) | 163 // optional, can be NULL, 0 (IN) |
| 147 // use_context -- whether to use the context value | 164 // use_context -- whether to use the context value |
| 148 // (needed to distinguish no context from | 165 // (needed to distinguish no context from |
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| 160 virtual bool SetDtlsSrtpCiphers(const std::vector<std::string>& ciphers); | 177 virtual bool SetDtlsSrtpCiphers(const std::vector<std::string>& ciphers); |
| 161 virtual bool GetDtlsSrtpCipher(std::string* cipher); | 178 virtual bool GetDtlsSrtpCipher(std::string* cipher); |
| 162 | 179 |
| 163 // Capabilities testing | 180 // Capabilities testing |
| 164 static bool HaveDtls(); | 181 static bool HaveDtls(); |
| 165 static bool HaveDtlsSrtp(); | 182 static bool HaveDtlsSrtp(); |
| 166 static bool HaveExporter(); | 183 static bool HaveExporter(); |
| 167 | 184 |
| 168 // Returns the default Ssl cipher used between streams of this class | 185 // Returns the default Ssl cipher used between streams of this class |
| 169 // for the given protocol version. This is used by the unit tests. | 186 // for the given protocol version. This is used by the unit tests. |
| 170 // TODO(torbjorng@webrtc.org): Fix callers to avoid default parameter. | 187 // TODO(guoweis): Move this away from a static class method. |
| 171 static std::string GetDefaultSslCipher(SSLProtocolVersion version, | 188 static uint16_t GetDefaultSslCipherForTest(SSLProtocolVersion version, |
| 172 KeyType key_type = KT_DEFAULT); | 189 KeyType key_type); |
| 190 |
| 191 // TODO(guoweis): Move this away from a static class method. Currently this is |
| 192 // introduced such that any caller could depend on sslstreamadapter.h without |
| 193 // depending on specific SSL implementation. |
| 194 static std::string GetSslCipherSuiteName(uint16_t cipher); |
| 173 | 195 |
| 174 private: | 196 private: |
| 175 // If true, the server certificate need not match the configured | 197 // If true, the server certificate need not match the configured |
| 176 // server_name, and in fact missing certificate authority and other | 198 // server_name, and in fact missing certificate authority and other |
| 177 // verification errors are ignored. | 199 // verification errors are ignored. |
| 178 bool ignore_bad_cert_; | 200 bool ignore_bad_cert_; |
| 179 | 201 |
| 180 // If true (default), the client is required to provide a certificate during | 202 // If true (default), the client is required to provide a certificate during |
| 181 // handshake. If no certificate is given, handshake fails. This applies to | 203 // handshake. If no certificate is given, handshake fails. This applies to |
| 182 // server mode only. | 204 // server mode only. |
| 183 bool client_auth_enabled_; | 205 bool client_auth_enabled_; |
| 184 }; | 206 }; |
| 185 | 207 |
| 186 } // namespace rtc | 208 } // namespace rtc |
| 187 | 209 |
| 188 #endif // WEBRTC_BASE_SSLSTREAMADAPTER_H_ | 210 #endif // WEBRTC_BASE_SSLSTREAMADAPTER_H_ |
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