Wire up RTP keep-alive in ortc api.

webrtc/call/call.h

 /*
  *  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.
  */
 #ifndef WEBRTC_CALL_CALL_H_
@@ skipping 187 matching lines @@
   virtual void OnTransportOverheadChanged(
       MediaType media,
       int transport_overhead_per_packet) = 0;
 
   virtual void OnNetworkRouteChanged(
       const std::string& transport_name,
       const rtc::NetworkRoute& network_route) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
+  // Try to update the RTP keep-alive part of the call config. This is allowed
+  // only before creating and send streams. Returns true on success.

[Taylor Brandstetter, 2017/07/12 16:19:50]

I think "and" was meant to be "any" here

[sprang_webrtc, 2017/07/16 09:34:03]

Done.

+  virtual bool SetRtpKeepAliveConfig(const RtpKeepAliveConfig& config) = 0;
+
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_CALL_H_