Wire up RTT statistics to webrtc::Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788

Review URL: https://webrtc-codereview.appspot.com/32249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7876 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/video/call.cc b/webrtc/video/call.cc
index 512e82c..15cc835 100644
--- a/webrtc/video/call.cc
+++ b/webrtc/video/call.cc
@@ -342,6 +342,9 @@
          ++it) {
       stats.pacer_delay_ms =
           std::max(it->second->GetPacerQueuingDelayMs(), stats.pacer_delay_ms);
+      int rtt_ms = it->second->GetRtt();
+      if (rtt_ms > 0)
+        stats.rtt_ms = rtt_ms;
     }
   }
   return stats;