Wire up RTP keep-alive in ortc api.

webrtc/test/call_test.h

Index: webrtc/test/call_test.h
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
index 11d0198cb0e1226733a7d46677e31092ee345540..5186afa7533881e625692f20f52962c30403a358 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -14,6 +14,7 @@
 #include <vector>
 
 #include "webrtc/call/call.h"
+#include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_audio_device.h"
@@ -108,6 +109,7 @@ class CallTest : public ::testing::Test {
 
   std::unique_ptr<webrtc::RtcEventLog> event_log_;
   std::unique_ptr<Call> sender_call_;
+  RtpTransportControllerSend* sender_call_transport_controller_;
   std::unique_ptr<PacketTransport> send_transport_;
   VideoSendStream::Config video_send_config_;
   VideoEncoderConfig video_encoder_config_;
@@ -182,6 +184,8 @@ class BaseTest : public RtpRtcpObserver {
 
   virtual Call::Config GetSenderCallConfig();
   virtual Call::Config GetReceiverCallConfig();
+  virtual void OnRtpTransportControllerSendCreated(
+      RtpTransportControllerSend* controller);
   virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
 
   virtual test::PacketTransport* CreateSendTransport(Call* sender_call);