Wire up RTP keep-alive in ortc api.

webrtc/ortc/rtptransportcontrolleradapter.cc

 /*
  *  Copyright 2017 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 234 matching lines @@
                                             cricket::CA_ANSWER, nullptr)) {
         break;
       }
     }
     return RTCError::OK();
   } while (false);
   LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
                        "Failed to apply new RTCP parameters.");
 }
 
+RTCError RtpTransportControllerAdapter::SetRtpTransportParameters(
+    const RtpTransportParameters& parameters) {
+  if (!call_->SetRtpKeepAliveConfig(parameters.keepalive)) {
+    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
+                         "Failed to apply new RTP transport parameters.");
+  }
+  return RTCError::OK();
+}
+
 RTCError RtpTransportControllerAdapter::ValidateAndApplyAudioSenderParameters(
     const RtpParameters& parameters,
     uint32_t* primary_ssrc) {
   RTC_DCHECK(voice_channel_);
   RTC_DCHECK(have_audio_sender_);
 
   auto codecs_result = ToCricketCodecs<cricket::AudioCodec>(parameters.codecs);
   if (!codecs_result.ok()) {
     return codecs_result.MoveError();
   }
@@ skipping 709 matching lines @@
     local_description->set_cryptos(cryptos);
 
     cryptos.clear();
     cryptos.push_back(*(rtp_transport->GetInternal()->send_key()));
     remote_description->set_cryptos(cryptos);
   }
   return RTCError::OK();
 }
 
 }  // namespace webrtc