Support encrypted RTP extensions (RFC 6904)

webrtc/rtc_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 60 matching lines @@
   CryptoOptions() {}
 
   // 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;
+
+  // If set to true, encrypted RTP header extensions as defined in RFC 6904
+  // will be negotiated. They will only be used if both peers support them.
+  bool enable_encrypted_rtp_header_extensions = 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
@@ skipping 175 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_RTC_BASE_SSLSTREAMADAPTER_H_