Pick the DTLS handshake timeout based on the ICE RTT estimate

webrtc/base/opensslstreamadapter.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 58 matching lines @@
       size_t digest_len,
       SSLPeerCertificateDigestError* error = nullptr) override;
 
   std::unique_ptr<SSLCertificate> GetPeerCertificate() const override;
 
   // Goes from state SSL_NONE to either SSL_CONNECTING or SSL_WAIT, depending
   // on whether the underlying stream is already open or not.
   int StartSSL() override;
   void SetMode(SSLMode mode) override;
   void SetMaxProtocolVersion(SSLProtocolVersion version) override;
+  void SetInitialRetransmissionTimeout(int timeout_ms) override;
 
   StreamResult Read(void* data,
                     size_t data_len,
                     size_t* read,
                     int* error) override;
   StreamResult Write(const void* data,
                      size_t data_len,
                      size_t* written,
                      int* error) override;
   void Close() override;
@@ skipping 116 matching lines @@
   std::string peer_certificate_digest_algorithm_;
 
   // The DtlsSrtp ciphers
   std::string srtp_ciphers_;
 
   // Do DTLS or not
   SSLMode ssl_mode_;
 
   // Max. allowed protocol version
   SSLProtocolVersion ssl_max_version_;
+
+  // A 50-ms initial timeout ensures rapid setup on fast connections, but may
+  // be too aggressive for low bandwidth links.
+  int dtls_handshake_timeout_ms_ = 50;
 };
 
 /////////////////////////////////////////////////////////////////////////////
 
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_OPENSSLSTREAMADAPTER_H__