Implement RTP keepalive in native stack.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 317 matching lines @@
 
   bool TimeToSendFullNackList(int64_t now) const;
 
   std::unique_ptr<RTPSender> rtp_sender_;
   RTCPSender rtcp_sender_;
   RTCPReceiver rtcp_receiver_;
 
   const Clock* const clock_;
 
   const bool audio_;
-  int64_t last_process_time_;
+
+  const RtpKeepAliveConfig 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
   int64_t nack_last_time_sent_full_;
   uint32_t nack_last_time_sent_full_prev_;
   uint16_t nack_last_seq_number_sent_;
 
   KeyFrameRequestMethod key_frame_req_method_;
 
   RemoteBitrateEstimator* remote_bitrate_;
 
   RtcpRttStats* rtt_stats_;
 
   PacketLossStats send_loss_stats_;
   PacketLossStats receive_loss_stats_;
 
   // The processed RTT from RtcpRttStats.
   rtc::CriticalSection critical_section_rtt_;
   int64_t rtt_ms_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_