Wire up RTP keep-alive in ortc api.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 194 matching lines @@
   void SignalChannelNetworkState(MediaType media, NetworkState state) override;
 
   void OnTransportOverheadChanged(MediaType media,
                                   int transport_overhead_per_packet) override;
 
   void OnNetworkRouteChanged(const std::string& transport_name,
                              const rtc::NetworkRoute& network_route) override;
 
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
-
   // Implements BitrateObserver.
   void OnNetworkChanged(uint32_t bitrate_bps,
                         uint8_t fraction_loss,
                         int64_t rtt_ms,
                         int64_t probing_interval_ms) override;
 
   // Implements BitrateAllocator::LimitObserver.
   void OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
                                  uint32_t max_padding_bitrate_bps) override;
 
@@ skipping 501 matching lines @@
   }
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
   // Copy ssrcs from |config| since |config| is moved.
   std::vector<uint32_t> ssrcs = config.rtp.ssrcs;
   VideoSendStream* send_stream = new VideoSendStream(
       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_);
     for (uint32_t ssrc : ssrcs) {
       RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
       video_send_ssrcs_[ssrc] = send_stream;
     }
     video_send_streams_.insert(send_stream);
   }
   send_stream->SignalNetworkState(video_network_state_);
@@ skipping 668 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc