Change initial DTLS retransmission timer from 1 second to 50ms.

webrtc/p2p/base/dtlstransportchannel_unittest.cc

 /*
  *  Copyright 2011 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.
  */
 
 #include <memory>
 #include <set>
 
 #include "webrtc/p2p/base/dtlstransport.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stringutils.h"
 
-#define MAYBE_SKIP_TEST(feature)                    \
-  if (!(rtc::SSLStreamAdapter::feature())) {  \
-    LOG(LS_INFO) << "Feature disabled... skipping"; \
-    return;                                         \
+#define MAYBE_SKIP_TEST(feature)                              \
+  if (!(rtc::SSLStreamAdapter::feature())) {                  \
+    LOG(LS_INFO) << #feature " feature disabled... skipping"; \
+    return;                                                   \
   }
 
 static const char kIceUfrag1[] = "TESTICEUFRAG0001";
 static const char kIcePwd1[] = "TESTICEPWD00000000000001";
 static const size_t kPacketNumOffset = 8;
 static const size_t kPacketHeaderLen = 12;
 static const int kFakePacketId = 0x1234;
 static const int kTimeout = 10000;
 
 static bool IsRtpLeadByte(uint8_t b) {
@@ skipping 367 matching lines @@
   size_t packet_size_ = 0u;
   std::set<int> received_;
   bool use_dtls_srtp_ = false;
   rtc::SSLProtocolVersion ssl_max_version_ = rtc::SSL_PROTOCOL_DTLS_12;
   bool negotiated_dtls_ = false;
   int received_dtls_client_hellos_ = 0;
   int received_dtls_server_hellos_ = 0;
   rtc::SentPacket sent_packet_;
 };
 
-
+// Note that this test always uses a FakeClock, due to the |fake_clock_| member
+// variable.
 class DtlsTransportChannelTest : public testing::Test {
  public:
   DtlsTransportChannelTest()
       : client1_("P1"),
         client2_("P2"),
         channel_ct_(1),
         use_dtls_(false),
         use_dtls_srtp_(false),
         ssl_expected_version_(rtc::SSL_PROTOCOL_DTLS_12) {}
 
@@ skipping 127 matching lines @@
   }
 
   void TestTransfer(size_t channel, size_t size, size_t count, bool srtp) {
     LOG(LS_INFO) << "Expect packets, size=" << size;
     client2_.ExpectPackets(channel, size);
     client1_.SendPackets(channel, size, count, srtp);
     EXPECT_EQ_WAIT(count, client2_.NumPacketsReceived(), kTimeout);
   }
 
  protected:
+  rtc::ScopedFakeClock fake_clock_;
   DtlsTestClient client1_;
   DtlsTestClient client2_;
   int channel_ct_;
   bool use_dtls_;
   bool use_dtls_srtp_;
   rtc::SSLProtocolVersion ssl_expected_version_;
 };
 
 // Test that transport negotiation of ICE, no DTLS works properly.
 TEST_F(DtlsTransportChannelTest, TestChannelSetupIce) {
@@ skipping 393 matching lines @@
   client1_.transport()->SetRemoteTransportDescription(
       MakeTransportDescription(client2_.certificate(),
                                cricket::CONNECTIONROLE_ACTIVE),
       cricket::CA_ANSWER, nullptr);
   EXPECT_TRUE(client1_.Connect(&client2_, true));
   EXPECT_TRUE_WAIT(
       client1_.all_channels_writable() && client2_.all_channels_writable(),
       kTimeout);
   EXPECT_EQ(1, client1_.received_dtls_client_hellos());
 }
+
+// Test that packets are retransmitted according to the expected schedule.
+// Each time a timeout occurs, the retransmission timer should be doubled up to
+// 60 seconds. The timer defaults to 1 second, but for WebRTC we should be
+// initializing it to 50ms.
+TEST_F(DtlsTransportChannelTest, TestRetransmissionSchedule) {
+  MAYBE_SKIP_TEST(HaveDtls);
+  // We can only change the retransmission schedule with a recently-added
+  // BoringSSL API. Skip the test if not built with BoringSSL.
+  MAYBE_SKIP_TEST(IsBoringSsl);
+
+  PrepareDtls(true, true, rtc::KT_DEFAULT);
+  // Exchange transport descriptions.
+  Negotiate(cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_ACTIVE);
+
+  // Make client2_ writable, but not client1_.
+  // This means client1_ will send DTLS client hellos but get no response.
+  EXPECT_TRUE(client2_.Connect(&client1_, true));
+  EXPECT_TRUE_WAIT(client2_.all_raw_channels_writable(), kTimeout);
+
+  // Wait for the first client hello to be sent.
+  EXPECT_EQ_WAIT(1, client1_.received_dtls_client_hellos(), kTimeout);
+  EXPECT_FALSE(client1_.all_raw_channels_writable());
+
+  static int timeout_schedule_ms[] = {50,   100,  200,   400,   800,   1600,
+                                      3200, 6400, 12800, 25600, 51200, 60000};
+
+  int expected_hellos = 1;
+  for (size_t i = 0;
+       i < (sizeof(timeout_schedule_ms) / sizeof(timeout_schedule_ms[0]));
+       ++i) {
+    // For each expected retransmission time, advance the fake clock a
+    // millisecond before the expected time and verify that no unexpected
+    // retransmissions were sent. Then advance it the final millisecond and
+    // verify that the expected retransmission was sent.
+    fake_clock_.AdvanceTime(
+        rtc::TimeDelta::FromMilliseconds(timeout_schedule_ms[i] - 1));
+    EXPECT_EQ(expected_hellos, client1_.received_dtls_client_hellos());
+    fake_clock_.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1));
+    EXPECT_EQ(++expected_hellos, client1_.received_dtls_client_hellos());
+  }
+}