Pick the DTLS handshake timeout based on the ICE RTT estimate

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 138 matching lines @@
   // Do DTLS or TLS.
   virtual void SetMode(SSLMode mode) = 0;
 
   // Set maximum supported protocol version. The highest version supported by
   // both ends will be used for the connection, i.e. if one party supports
   // DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
   // If requested version is not supported by underlying crypto library, the
   // next lower will be used.
   virtual void SetMaxProtocolVersion(SSLProtocolVersion version) = 0;
 
+  // Set the initial timeout for the handshake message. When the timeout
+  // expires, the handshake message gets retransmitted and the timeout is
+  // exponentially increased.
+  virtual void SetHandshakeTimeout(int timeout_ms) = 0;

[Taylor Brandstetter, 2017/02/02 18:07:41]

nit: Better name would be "InitialRetransmissionTimeout"

[pthatcher2, 2017/02/02 20:09:44]

+1

[skvlad, 2017/02/02 22:19:33]

I made it InitialHandshakeRetransmissionTimeout. Long but accurate.

[Taylor Brandstetter, 2017/02/02 22:57:27]

I'd still call it just "InitialRetransmissionTimeout", since in theory it
applies to any DTLS messages (not just the handshake).

+
   // StartSSL starts negotiation with a peer, whose certificate is verified
   // using the certificate digest. Generally, SetIdentity() and possibly
   // SetServerRole() should have been called before this.
   // SetPeerCertificateDigest() must also be called. It may be called after
   // StartSSLWithPeer() but must be called before the underlying stream opens.
   //
   // Use of the stream prior to calling StartSSL will pass data in clear text.
   // Calling StartSSL causes SSL negotiation to begin as soon as possible: right
   // away if the underlying wrapped stream is already opened, or else as soon as
   // it opens.
@@ skipping 88 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_