Adding test for unordered, fragmented SCTP message delivery.

webrtc/api/peerconnection_unittest.cc

 /*
  *  Copyright 2012 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 13 matching lines @@
 #include "webrtc/api/peerconnection.h"
 #include "webrtc/api/peerconnectionfactory.h"
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/test/fakeaudiocapturemodule.h"
 #include "webrtc/api/test/fakeconstraints.h"
 #include "webrtc/api/test/fakeperiodicvideocapturer.h"
 #include "webrtc/api/test/fakertccertificategenerator.h"
 #include "webrtc/api/test/fakevideotrackrenderer.h"
 #include "webrtc/api/test/mockpeerconnectionobservers.h"
 #include "webrtc/base/gunit.h"
+#include "webrtc/base/helpers.h"
 #include "webrtc/base/physicalsocketserver.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/virtualsocketserver.h"
 #include "webrtc/media/engine/fakewebrtcvideoengine.h"
 #include "webrtc/p2p/base/fakeportallocator.h"
 #include "webrtc/p2p/base/p2pconstants.h"
 #include "webrtc/p2p/base/sessiondescription.h"
 #include "webrtc/pc/mediasession.h"
@@ skipping 1286 matching lines @@
   void SendRtpData(webrtc::DataChannelInterface* dc, const std::string& data) {
     // Messages may get lost on the unreliable DataChannel, so we send multiple
     // times to avoid test flakiness.
     static const size_t kSendAttempts = 5;
 
     for (size_t i = 0; i < kSendAttempts; ++i) {
       dc->Send(DataBuffer(data));
     }
   }
 
+  rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); }
+
   PeerConnectionTestClient* initializing_client() {
     return initiating_client_.get();
   }
 
   // Set the |initiating_client_| to the |client| passed in and return the
   // original |initiating_client_|.
   PeerConnectionTestClient* set_initializing_client(
       PeerConnectionTestClient* client) {
     PeerConnectionTestClient* old = initiating_client_.release();
     initiating_client_.reset(client);
@@ skipping 572 matching lines @@
   receiving_client()->data_channel()->Send(DataBuffer(data));
   EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
                  kMaxWaitMs);
 
   receiving_client()->data_channel()->Close();
   EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(),
                    kMaxWaitMs);
   EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
 }
 
+TEST_F(P2PTestConductor, UnorderedSctpDataChannel) {
+  ASSERT_TRUE(CreateTestClients());
+  webrtc::DataChannelInit init;
+  init.ordered = false;
+  initializing_client()->CreateDataChannel(&init);
+
+  // Introduce random network delays.
+  // Otherwise it's not a true "unordered" test.
+  virtual_socket_server()->set_delay_mean(20);
+  virtual_socket_server()->set_delay_stddev(5);
+  virtual_socket_server()->UpdateDelayDistribution();
+
+  initializing_client()->Negotiate();
+  ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
+  EXPECT_TRUE_WAIT(receiving_client()->data_channel() != nullptr, kMaxWaitMs);
+  EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
+                   kMaxWaitMs);
+  EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
+
+  static constexpr int kNumMessages = 100;
+  // Deliberately chosen to be larger than the MTU so messages get fragmented.
+  static constexpr size_t kMaxMessageSize = 4096;
+  // Create and send random messages.
+  std::vector<std::string> sent_messages;
+  for (int i = 0; i < kNumMessages; ++i) {
+    size_t length = (rand() % kMaxMessageSize) + 1;
+    std::string message;
+    ASSERT_TRUE(rtc::CreateRandomString(length, &message));
+    initializing_client()->data_channel()->Send(DataBuffer(message));
+    receiving_client()->data_channel()->Send(DataBuffer(message));
+    sent_messages.push_back(message);
+  }
+
+  EXPECT_EQ_WAIT(
+      kNumMessages,
+      initializing_client()->data_observer()->received_message_count(),
+      kMaxWaitMs);
+  EXPECT_EQ_WAIT(kNumMessages,
+                 receiving_client()->data_observer()->received_message_count(),
+                 kMaxWaitMs);
+
+  // Sort and compare to make sure none of the messages were corrupted.
+  std::vector<std::string> initializing_client_received_messages =
+      initializing_client()->data_observer()->messages();
+  std::vector<std::string> receiving_client_received_messages =
+      receiving_client()->data_observer()->messages();
+  std::sort(sent_messages.begin(), sent_messages.end());
+  std::sort(initializing_client_received_messages.begin(),
+            initializing_client_received_messages.end());
+  std::sort(receiving_client_received_messages.begin(),
+            receiving_client_received_messages.end());
+  EXPECT_EQ(sent_messages, initializing_client_received_messages);
+  EXPECT_EQ(sent_messages, receiving_client_received_messages);
+
+  receiving_client()->data_channel()->Close();
+  EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(),
+                   kMaxWaitMs);
+  EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
+}
+
 // This test sets up a call between two parties and creates a data channel.
 // The test tests that received data is buffered unless an observer has been
 // registered.
 // Rtp data channels can receive data before the underlying
 // transport has detected that a channel is writable and thus data can be
 // received before the data channel state changes to open. That is hard to test
 // but the same buffering is used in that case.
 TEST_F(P2PTestConductor, RegisterDataChannelObserver) {
   FakeConstraints setup_constraints;
   setup_constraints.SetAllowRtpDataChannels();
@@ skipping 475 matching lines @@
   server.urls.push_back("turn:hostname2");
   servers.push_back(server);
   EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
   EXPECT_EQ(2U, turn_servers_.size());
   EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
 }
 
 #endif // if !defined(THREAD_SANITIZER)
 
 }  // namespace