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

webrtc/base/sslstreamadapter.h

Index: webrtc/base/sslstreamadapter.h
diff --git a/webrtc/base/sslstreamadapter.h b/webrtc/base/sslstreamadapter.h
index e1d1d852156786f5e84e202574ba4dae50a3ed1c..62a724996ef621cc7ea368c30a1589a28ee07bb8 100644
--- a/webrtc/base/sslstreamadapter.h
+++ b/webrtc/base/sslstreamadapter.h
@@ -38,9 +38,7 @@ const int SRTP_AEAD_AES_128_GCM = 0x0007;
 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.
@@ -82,6 +80,11 @@ struct CryptoOptions {
   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