Negotiate the same SRTP crypto suites for every DTLS association formed.

webrtc/pc/peerconnection_integrationtest.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 2002 matching lines @@
 // Test that GCM isn't used if only the answerer supports it.
 TEST_F(PeerConnectionIntegrationTest,
        NonGcmCipherUsedWhenOnlyCalleeSupportsGcm) {
   bool local_gcm_enabled = false;
   bool remote_gcm_enabled = true;
   int expected_cipher_suite = kDefaultSrtpCryptoSuite;
   TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
                                     expected_cipher_suite);
 }
 
+// Verify that media can be transmitted end-to-end when GCM crypto suites are
+// enabled. Note that the above tests, such as GcmCipherUsedWhenGcmSupported,
+// only verify that a GCM cipher is negotiated, and not necessarily that SRTP
+// works with it.
+TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) {
+  PeerConnectionFactory::Options gcm_options;
+  gcm_options.crypto_options.enable_gcm_crypto_suites = true;
+  ASSERT_TRUE(
+      CreatePeerConnectionWrappersWithOptions(gcm_options, gcm_options));
+  ConnectFakeSignaling();
+  // Do normal offer/answer and wait for some frames to be received in each
+  // direction.
+  caller()->AddAudioVideoMediaStream();
+  callee()->AddAudioVideoMediaStream();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ExpectNewFramesReceivedWithWait(
+      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
+      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
+      kMaxWaitForFramesMs);
+}
+
 // This test sets up a call between two parties with audio, video and an RTP
 // data channel.
 TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithRtpDataChannel) {
   FakeConstraints setup_constraints;
   setup_constraints.SetAllowRtpDataChannels();
   ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints(&setup_constraints,
                                                           &setup_constraints));
   ConnectFakeSignaling();
   // Expect that data channel created on caller side will show up for callee as
   // well.
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   // Ensure data can be sent in both directions.
   std::string data = "hello world";
   caller()->data_channel()->Send(DataBuffer(data));
   EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
                  kDefaultTimeout);
   callee()->data_channel()->Send(DataBuffer(data));
   EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
                  kDefaultTimeout);
 }
 
+// Set up a connection initially just using SCTP data channels, later upgrading
+// to audio/video, ensuring frames are received end-to-end. Effectively the
+// inverse of the test above.
+// This was broken in M57; see https://crbug.com/711243
+TEST_F(PeerConnectionIntegrationTest, SctpDataChannelToAudioVideoUpgrade) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  // Do initial offer/answer with just data channel.
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Wait until data can be sent over the data channel.
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  // Do subsequent offer/answer with two-way audio and video. Audio and video
+  // should end up bundled on the DTLS/ICE transport already used for data.
+  caller()->AddAudioVideoMediaStream();
+  callee()->AddAudioVideoMediaStream();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ExpectNewFramesReceivedWithWait(
+      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
+      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
+      kMaxWaitForFramesMs);
+}
+
 #endif  // HAVE_SCTP
 
 // Test that the ICE connection and gathering states eventually reach
 // "complete".
 TEST_F(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Do normal offer/answer.
   caller()->AddAudioVideoMediaStream();
   callee()->AddAudioVideoMediaStream();
@@ skipping 420 matching lines @@
 
   auto contributing_sources = receiver->GetSources();
   ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
   EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
             contributing_sources[0].source_id());
 }
 
 }  // namespace
 
 #endif  // if !defined(THREAD_SANITIZER)