Move RTP keep-alive config from VideoSendStream::Config to Call::Config

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 41f53feee7c57ed5e42753ad9e2a78075eb8a22c..3966d5e0309add865df305fbf284f689c317400e 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -727,7 +727,8 @@ 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_);
+      std::move(encoder_config), suspended_video_send_ssrcs_,
+      config_.keepalive_config);
 
   {
     WriteLockScoped write_lock(*send_crit_);