Wire up RTP keep-alive in ortc api.

webrtc/ortc/ortcfactory_integrationtest.cc

 /*
  *  Copyright 2017 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 110 matching lines @@
     return {ortc_factory1_->CreateRtpTransportController().MoveValue(),
             ortc_factory2_->CreateRtpTransportController().MoveValue()};
   }
 
   // Helper function that creates a pair of RtpTransports between
   // |ortc_factory1_| and |ortc_factory2_|. Expected to be called with the
   // result of CreateAndConnectUdpTransportPair. |rtcp_udp_transports| can be
   // empty if RTCP muxing is used. |transport_controllers| can be empty if
   // these transports are being created using a default transport controller.
   RtpTransportPair CreateRtpTransportPair(
-      const RtcpParameters& rtcp_parameters,
+      const RtpTransportParameters& parameters,
       const UdpTransportPair& rtp_udp_transports,
       const UdpTransportPair& rtcp_udp_transports,
       const RtpTransportControllerPair& transport_controllers) {
     auto transport_result1 = ortc_factory1_->CreateRtpTransport(
-        rtcp_parameters, rtp_udp_transports.first.get(),
+        parameters, rtp_udp_transports.first.get(),
         rtcp_udp_transports.first.get(), transport_controllers.first.get());
     auto transport_result2 = ortc_factory2_->CreateRtpTransport(
-        rtcp_parameters, rtp_udp_transports.second.get(),
+        parameters, rtp_udp_transports.second.get(),
         rtcp_udp_transports.second.get(), transport_controllers.second.get());
     return {transport_result1.MoveValue(), transport_result2.MoveValue()};
   }
 
   SrtpTransportPair CreateSrtpTransportPair(
-      const RtcpParameters& rtcp_parameters,
+      const RtpTransportParameters& parameters,
       const UdpTransportPair& rtp_udp_transports,
       const UdpTransportPair& rtcp_udp_transports,
       const RtpTransportControllerPair& transport_controllers) {
     auto transport_result1 = ortc_factory1_->CreateSrtpTransport(
-        rtcp_parameters, rtp_udp_transports.first.get(),
+        parameters, rtp_udp_transports.first.get(),
         rtcp_udp_transports.first.get(), transport_controllers.first.get());
     auto transport_result2 = ortc_factory2_->CreateSrtpTransport(
-        rtcp_parameters, rtp_udp_transports.second.get(),
+        parameters, rtp_udp_transports.second.get(),
         rtcp_udp_transports.second.get(), transport_controllers.second.get());
     return {transport_result1.MoveValue(), transport_result2.MoveValue()};
   }
 
   // For convenience when |rtcp_udp_transports| and |transport_controllers|
   // aren't needed.
   RtpTransportPair CreateRtpTransportPair(
-      const RtcpParameters& rtcp_parameters,
+      const RtpTransportParameters& parameters,
       const UdpTransportPair& rtp_udp_transports) {
-    return CreateRtpTransportPair(rtcp_parameters, rtp_udp_transports,
+    return CreateRtpTransportPair(parameters, rtp_udp_transports,
                                   UdpTransportPair(),
                                   RtpTransportControllerPair());
   }
 
   SrtpTransportPair CreateSrtpTransportPairAndSetKeys(
-      const RtcpParameters& rtcp_parameters,
+      const RtpTransportParameters& parameters,
       const UdpTransportPair& rtp_udp_transports) {
     SrtpTransportPair srtp_transports = CreateSrtpTransportPair(
-        rtcp_parameters, rtp_udp_transports, UdpTransportPair(),
+        parameters, rtp_udp_transports, UdpTransportPair(),
         RtpTransportControllerPair());
     EXPECT_TRUE(srtp_transports.first->SetSrtpSendKey(kTestCryptoParams1).ok());
     EXPECT_TRUE(
         srtp_transports.first->SetSrtpReceiveKey(kTestCryptoParams2).ok());
     EXPECT_TRUE(
         srtp_transports.second->SetSrtpSendKey(kTestCryptoParams2).ok());
     EXPECT_TRUE(
         srtp_transports.second->SetSrtpReceiveKey(kTestCryptoParams1).ok());
     return srtp_transports;
   }
 
   SrtpTransportPair CreateSrtpTransportPairAndSetMismatchingKeys(
-      const RtcpParameters& rtcp_parameters,
+      const RtpTransportParameters& parameters,
       const UdpTransportPair& rtp_udp_transports) {
     SrtpTransportPair srtp_transports = CreateSrtpTransportPair(
-        rtcp_parameters, rtp_udp_transports, UdpTransportPair(),
+        parameters, rtp_udp_transports, UdpTransportPair(),
         RtpTransportControllerPair());
     EXPECT_TRUE(srtp_transports.first->SetSrtpSendKey(kTestCryptoParams1).ok());
     EXPECT_TRUE(
         srtp_transports.first->SetSrtpReceiveKey(kTestCryptoParams2).ok());
     EXPECT_TRUE(
         srtp_transports.second->SetSrtpSendKey(kTestCryptoParams1).ok());
     EXPECT_TRUE(
         srtp_transports.second->SetSrtpReceiveKey(kTestCryptoParams2).ok());
     return srtp_transports;
   }
@@ skipping 352 matching lines @@
   // RTP/RTCP.
   auto audio_rtp_udp_transports = CreateAndConnectUdpTransportPair();
   auto audio_rtcp_udp_transports = CreateAndConnectUdpTransportPair();
   auto video_rtp_udp_transports = CreateAndConnectUdpTransportPair();
   auto video_rtcp_udp_transports = CreateAndConnectUdpTransportPair();
 
   // Since we have multiple RTP transports on each side, we need an RTP
   // transport controller.
   auto transport_controllers = CreateRtpTransportControllerPair();
 
-  RtcpParameters audio_rtcp_parameters;
-  audio_rtcp_parameters.mux = false;
-  auto audio_srtp_transports =
-      CreateSrtpTransportPair(audio_rtcp_parameters, audio_rtp_udp_transports,
-                              audio_rtcp_udp_transports, transport_controllers);
+  RtpTransportParameters audio_rtp_transport_parameters;
+  audio_rtp_transport_parameters.rtcp.mux = false;
+  auto audio_srtp_transports = CreateSrtpTransportPair(
+      audio_rtp_transport_parameters, audio_rtp_udp_transports,
+      audio_rtcp_udp_transports, transport_controllers);
 
-  RtcpParameters video_rtcp_parameters;
-  video_rtcp_parameters.mux = false;
-  video_rtcp_parameters.reduced_size = true;
-  auto video_srtp_transports =
-      CreateSrtpTransportPair(video_rtcp_parameters, video_rtp_udp_transports,
-                              video_rtcp_udp_transports, transport_controllers);
+  RtpTransportParameters video_rtp_transport_parameters;
+  video_rtp_transport_parameters.rtcp.mux = false;
+  video_rtp_transport_parameters.rtcp.reduced_size = true;
+  auto video_srtp_transports = CreateSrtpTransportPair(
+      video_rtp_transport_parameters, video_rtp_udp_transports,
+      video_rtcp_udp_transports, transport_controllers);
 
   // Set keys for SRTP transports.
   audio_srtp_transports.first->SetSrtpSendKey(kTestCryptoParams1);
   audio_srtp_transports.first->SetSrtpReceiveKey(kTestCryptoParams2);
   video_srtp_transports.first->SetSrtpSendKey(kTestCryptoParams3);
   video_srtp_transports.first->SetSrtpReceiveKey(kTestCryptoParams4);
 
   audio_srtp_transports.second->SetSrtpSendKey(kTestCryptoParams2);
   audio_srtp_transports.second->SetSrtpReceiveKey(kTestCryptoParams1);
   video_srtp_transports.second->SetSrtpSendKey(kTestCryptoParams4);
@@ skipping 99 matching lines @@
 // BaseChannel model relies on there being a single VoiceChannel and
 // VideoChannel, and these only support a single set of codecs/etc. per
 // send/receive direction.
 
 // TODO(deadbeef): End-to-end test for simulcast, once that's supported by this
 // API.
 
 #endif  // if !defined(THREAD_SANITIZER)
 
 }  // namespace webrtc