Implement RTP keepalive in native stack.

webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index 6fc6b862b85e8d91aa0d9453e678688bf4fb36a2..9eb37a6efd39875ff78cad686bc17515c354fa97 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -22,6 +22,7 @@
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
@@ -92,6 +93,7 @@ class RtpRtcp : public Module {
     SendPacketObserver* send_packet_observer = nullptr;
     RateLimiter* retransmission_rate_limiter = nullptr;
     OverheadObserver* overhead_observer = nullptr;
+    VideoSendStream::Config::Rtp::KeepAlive keepalive_config;
 
    private:
     RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);