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 97 matching lines @@
 }
 
 PacketTransportInterface* RtpTransport::GetRtpPacketTransport() const {
   return rtp_packet_transport_;
 }
 
 PacketTransportInterface* RtpTransport::GetRtcpPacketTransport() const {
   return rtcp_packet_transport_;
 }
 
-RTCError RtpTransport::SetRtcpParameters(const RtcpParameters& parameters) {
-  if (rtcp_parameters_.mux && !parameters.mux) {
+RTCError RtpTransport::SetParameters(const RtpTransportParameters& parameters) {
+  if (parameters_.rtcp.mux && !parameters.rtcp.mux) {
     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
                          "Disabling RTCP muxing is not allowed.");
   }
-
-  RtcpParameters new_parameters = parameters;
-
-  if (new_parameters.cname.empty()) {
-    new_parameters.cname = rtcp_parameters_.cname;
+  if (parameters.keepalive != parameters_.keepalive) {
+    LOG_AND_RETURN_ERROR(
+        RTCErrorType::INVALID_MODIFICATION,
+        "RTP keep-alive parameters not supported by this channel.");

[Taylor Brandstetter, 2017/08/03 01:18:04]

Why is this message changing?

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

This is a new message. I just verify that you don't try to set keep-alive
config, since it's not wired up with this implementation.

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

Sorry, I thought this was RtpTransportAdapter when reviewing, carry on.

   }
 
-  rtcp_parameters_ = new_parameters;
+  RtpTransportParameters new_parameters = parameters;
+
+  if (new_parameters.rtcp.cname.empty()) {
+    new_parameters.rtcp.cname = parameters_.rtcp.cname;
+  }
+
+  parameters_ = new_parameters;
   return RTCError::OK();
 }
 
-RtcpParameters RtpTransport::GetRtcpParameters() const {
-  return rtcp_parameters_;
+RtpTransportParameters RtpTransport::GetParameters() const {
+  return parameters_;
 }
 
 RtpTransportAdapter* RtpTransport::GetInternal() {
   return nullptr;
 }
 
 void RtpTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) {
   SetReadyToSend(transport == rtcp_packet_transport_, true);
 }
 
@@ skipping 57 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