Implement RTP keepalive in native stack.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 0c46e40e1ec38d8414679cde93de00c1ea2c4789..06924a32ff450c5dcb31258c454cf4fed9a21795 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -25,6 +25,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
+#include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
@@ -335,9 +336,12 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
   const Clock* const clock_;
 
   const bool audio_;
-  int64_t last_process_time_;
+
+  const VideoSendStream::Config::Rtp::KeepAlive keepalive_config_;
   int64_t last_bitrate_process_time_;
   int64_t last_rtt_process_time_;
+  int64_t next_process_time_;
+  int64_t next_keepalive_time_;
   uint16_t packet_overhead_;
 
   // Send side