Wire up RTP keep-alive in ortc api.

webrtc/api/ortc/ortcfactoryinterface.h

 /*
  *  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 95 matching lines @@
   // TODO(deadbeef): Add MediaConfig and RtcEventLog arguments?
   virtual RTCErrorOr<std::unique_ptr<RtpTransportControllerInterface>>
   CreateRtpTransportController() = 0;
 
   // Creates an RTP transport using the provided packet transports and
   // transport controller.
   //
   // |rtp| will be used for sending RTP packets, and |rtcp| for RTCP packets.
   //
   // |rtp| can't be null. |rtcp| must be non-null if and only if
-  // |rtcp_parameters.mux| is false, indicating that RTCP muxing isn't used.
+  // |rtp_parameters.rtcp.mux| is false, indicating that RTCP muxing isn't used.
   // Note that if RTCP muxing isn't enabled initially, it can still enabled
-  // later through SetRtcpParameters.
+  // later through SetParameters.
   //
   // If |transport_controller| is null, one will automatically be created, and
   // its lifetime managed by the returned RtpTransport. This should only be
   // done if a single RtpTransport is being used to communicate with the remote
   // endpoint.
   virtual RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateRtpTransport(
-      const RtcpParameters& rtcp_parameters,
+      const RtpTransportParameters& rtp_parameters,
       PacketTransportInterface* rtp,
       PacketTransportInterface* rtcp,
       RtpTransportControllerInterface* transport_controller) = 0;
 
   // Creates an SrtpTransport which is an RTP transport that uses SRTP.
   virtual RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>
   CreateSrtpTransport(
-      const RtcpParameters& rtcp_parameters,
+      const RtpTransportParameters& rtp_parameters,
       PacketTransportInterface* rtp,
       PacketTransportInterface* rtcp,
       RtpTransportControllerInterface* transport_controller) = 0;
 
   // Returns the capabilities of an RTP sender of type |kind|. These
   // capabilities can be used to determine what RtpParameters to use to create
   // an RtpSender.
   //
   // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
   virtual RtpCapabilities GetRtpSenderCapabilities(
@@ skipping 86 matching lines @@
 
   // Creates an new local audio track wrapping |source|.
   virtual rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
       const std::string& id,
       AudioSourceInterface* source) = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_ORTC_ORTCFACTORYINTERFACE_H_