Wire up RTP keep-alive in ortc api.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 25dcf32a620c48965c22077d8c6d35cc3b298267..61ed66cf43ac36f274d02fc1dfb1616cded23694 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -212,7 +212,6 @@ class Call : public webrtc::Call,
 
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
-
   // Implements BitrateObserver.
   void OnNetworkChanged(uint32_t bitrate_bps,
                         uint8_t fraction_loss,
@@ -734,8 +733,7 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
       num_cpu_cores_, module_process_thread_.get(), &worker_queue_,
       call_stats_.get(), transport_send_.get(), bitrate_allocator_.get(),
       video_send_delay_stats_.get(), event_log_, std::move(config),
-      std::move(encoder_config), suspended_video_send_ssrcs_,
-      config_.keepalive_config);
+      std::move(encoder_config), suspended_video_send_ssrcs_);
 
   {
     WriteLockScoped write_lock(*send_crit_);