Wire up RTP keep-alive in ortc api.

webrtc/pc/rtptransport.h

Index: webrtc/pc/rtptransport.h
diff --git a/webrtc/pc/rtptransport.h b/webrtc/pc/rtptransport.h
index a86fa126bb8569cdaaafda36a0064285542e5fd4..0e4558e52fe8115a79ce552f080e2a9745d17f3a 100644
--- a/webrtc/pc/rtptransport.h
+++ b/webrtc/pc/rtptransport.h
@@ -51,8 +51,8 @@ class RtpTransport : public RtpTransportInterface, public sigslot::has_slots<> {
   PacketTransportInterface* GetRtcpPacketTransport() const override;
 
   // TODO(zstein): Use these RtcpParameters for configuration elsewhere.
-  RTCError SetRtcpParameters(const RtcpParameters& parameters) override;
-  RtcpParameters GetRtcpParameters() const override;
+  RTCError SetParameters(const RtpTransportParameters& parameters) override;
+  RtpTransportParameters GetParameters() const override;
 
   // Called whenever a transport's ready-to-send state changes. The argument
   // is true if all used transports are ready to send. This is more specific
@@ -108,7 +108,7 @@ class RtpTransport : public RtpTransportInterface, public sigslot::has_slots<> {
   bool rtp_ready_to_send_ = false;
   bool rtcp_ready_to_send_ = false;
 
-  RtcpParameters rtcp_parameters_;
+  RtpTransportParameters parameters_;
 
   cricket::BundleFilter bundle_filter_;
 };