Wire up RTP keep-alive in ortc api.

webrtc/ortc/ortcfactory.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 25 matching lines @@
       rtc::Thread* network_thread,
       rtc::Thread* signaling_thread,
       rtc::NetworkManager* network_manager,
       rtc::PacketSocketFactory* socket_factory,
       AudioDeviceModule* adm,
       std::unique_ptr<cricket::MediaEngineInterface> media_engine);
 
   RTCErrorOr<std::unique_ptr<RtpTransportControllerInterface>>
   CreateRtpTransportController() override;
 
+  RTCErrorOr<std::unique_ptr<RtpTransportControllerInterface>>
+  CreateRtpTransportController(const RtpTransportParameters& parameters);

[Taylor Brandstetter, 2017/08/03 01:18:03]

We shouldn't add this method; the fact that RtpTransportControllerAdapter knows
about RtpTransportParameters is an implementation detail. From an API
perspective, each RtpTransport has its own RtpTransportParameters, just like how
each RtpSender/RtpReceiver has its own RtpParameters.

[sprang_webrtc, 2017/08/03 13:08:14]

True, that's why I didn't add it to the interface. What if we just make it
private? I did this since it allows setting the call config on construction.

[Taylor Brandstetter, 2017/08/03 16:50:36]

I don't see that allowing setting the call config on construction buys much,
since you need to be able to set it after construction anyway, in the normal
case where the RtpTransport is created after the RtpTransportController. I also
can't find where this constructor is called (or even defined).

+
   RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateRtpTransport(
-      const RtcpParameters& rtcp_parameters,
+      const RtpTransportParameters& parameters,
       PacketTransportInterface* rtp,
       PacketTransportInterface* rtcp,
       RtpTransportControllerInterface* transport_controller) override;
 
   RTCErrorOr<std::unique_ptr<SrtpTransportInterface>> CreateSrtpTransport(
-      const RtcpParameters& rtcp_parameters,
+      const RtpTransportParameters& parameters,
       PacketTransportInterface* rtp,
       PacketTransportInterface* rtcp,
       RtpTransportControllerInterface* transport_controller) override;
 
   RtpCapabilities GetRtpSenderCapabilities(
       cricket::MediaType kind) const override;
 
   RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>> CreateRtpSender(
       rtc::scoped_refptr<MediaStreamTrackInterface> track,
       RtpTransportInterface* transport) override;
@@ skipping 78 matching lines @@
   std::string default_cname_;
 
   friend class OrtcFactoryInterface;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(OrtcFactory);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_ORTC_ORTCFACTORY_H_