Add support for GCM cipher suites from RFC 7714.

webrtc/base/sslstreamadapter.h

 /*
  *  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 WEBRTC_BASE_SSLSTREAMADAPTER_H_
 #define WEBRTC_BASE_SSLSTREAMADAPTER_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/base/stream.h"
 #include "webrtc/base/sslidentity.h"
 
 namespace rtc {
 
 // Constants for SSL profile.
 const int TLS_NULL_WITH_NULL_NULL = 0;
 
 // Constants for SRTP profiles.
 const int SRTP_INVALID_CRYPTO_SUITE = 0;
 const int SRTP_AES128_CM_SHA1_80 = 0x0001;
 const int SRTP_AES128_CM_SHA1_32 = 0x0002;
+const int SRTP_AEAD_AES_128_GCM = 0x0007;
+const int SRTP_AEAD_AES_256_GCM = 0x0008;
 
 // Cipher suite to use for SRTP. Typically a 80-bit HMAC will be used, except
 // in applications (voice) where the additional bandwidth may be significant.
 // A 80-bit HMAC is always used for SRTCP.
 // 128-bit AES with 80-bit SHA-1 HMAC.
 extern const char CS_AES_CM_128_HMAC_SHA1_80[];
 // 128-bit AES with 32-bit SHA-1 HMAC.
 extern const char CS_AES_CM_128_HMAC_SHA1_32[];
+// 128-bit AES GCM with 16 byte AEAD auth tag.
+extern const char CS_AEAD_AES_128_GCM[];
+// 256-bit AES GCM with 16 byte AEAD auth tag.
+extern const char CS_AEAD_AES_256_GCM[];
 
 // 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(const std::string& crypto_suite);
 
+// Get key length and salt length for given crypto suite. Returns true for
+// valid suites, otherwise false.
+bool SrtpCryptoSuiteParams(int crypto_suite, int *key_length, int *salt_length);
+
 // 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.
 //
@@ skipping 152 matching lines @@
 
   // 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_;
 };
 
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_SSLSTREAMADAPTER_H_