Wire up RTP keep-alive in ortc api.

webrtc/pc/rtptransport.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 117 matching lines @@
   }
 
   rtcp_parameters_ = new_parameters;
   return RTCError::OK();
 }
 
 RtcpParameters RtpTransport::GetRtcpParameters() const {
   return rtcp_parameters_;
 }
 
+RTCError RtpTransport::SetRtpTransportParameters(
+    const RtpTransportParameters& parameters) {
+  if (rtp_transport_parameters_.keepalive.timeout_interval_ms > 0 &&

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

The parameters in this class don't appear to be used. So it may be better to
leave this method as a TODO and just return an error, so people don't wrongly
assume things are wired up already.

Note that this class is currently only used in the PeerConnection object model
(hence it being in "pc/"), but it's what we plan to eventually use for both the
ORTC and PeerConnection code paths, replacing "RtpTransportAdapter" with it.

[ilnik, 2017/07/21 12:21:19]

Acknowledged.

+      rtp_transport_parameters_.keepalive != parameters.keepalive) {
+    LOG_AND_RETURN_ERROR(
+        RTCErrorType::INVALID_STATE,
+        "Cannot change RTP keep-alive parameters once enabled.");
+  }
+
+  rtp_transport_parameters_ = parameters;
+  return RTCError::OK();
+}
+
+RtpTransportParameters RtpTransport::GetRtpTransportParameters() const {
+  return rtp_transport_parameters_;
+}
+
 RtpTransportAdapter* RtpTransport::GetInternal() {
   return nullptr;
 }
 
 void RtpTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) {
   SetReadyToSend(transport == rtcp_packet_transport_, true);
 }
 
 void RtpTransport::SetReadyToSend(bool rtcp, bool ready) {
   if (rtcp) {
@@ skipping 55 matching lines @@
   }
   if (rtcp) {
     // Permit all (seemingly valid) RTCP packets.
     return true;
   }
   // Check whether we handle this payload.
   return HandlesPacket(packet->data(), packet->size());
 }
 
 }  // namespace webrtc