Negotiate the same SRTP crypto suites for every DTLS association formed.

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.
  */
 
@@ skipping 20 matching lines @@
 #ifndef SRTP_AES128_CM_SHA1_32
 const int SRTP_AES128_CM_SHA1_32 = 0x0002;
 #endif
 #ifndef SRTP_AEAD_AES_128_GCM
 const int SRTP_AEAD_AES_128_GCM = 0x0007;
 #endif
 #ifndef SRTP_AEAD_AES_256_GCM
 const int SRTP_AEAD_AES_256_GCM = 0x0008;
 #endif
 
-// 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.
+// Names of SRTP profiles listed above.
 // 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
@@ skipping 21 matching lines @@
   // Helper method to return an instance of the CryptoOptions with GCM crypto
   // suites disabled. This method should be used instead of depending on current
   // default values set by the constructor.
   static CryptoOptions NoGcm();
 
   // Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
   // if both sides enable it.
   bool enable_gcm_crypto_suites = false;
 };
 
+// Returns supported crypto suites, given |crypto_options|.
+// CS_AES_CM_128_HMAC_SHA1_32 will be preferred by default.
+std::vector<int> GetSupportedDtlsSrtpCryptoSuites(
+    const rtc::CryptoOptions& crypto_options);
+
 // 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 168 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_