diff --git a/talk/media/webrtc/webrtcmediaengine.cc b/talk/media/webrtc/webrtcmediaengine.cc
index 252b4e6070..67bea0cbc0 100644
--- a/talk/media/webrtc/webrtcmediaengine.cc
+++ b/talk/media/webrtc/webrtcmediaengine.cc
@@ -50,7 +50,6 @@ class WebRtcMediaEngine :
       WebRtcVideoDecoderFactory* decoder_factory) {
     voice_.SetAudioDeviceModule(adm, adm_sc);
     video_.SetVoiceEngine(&voice_);
-    video_.EnableTimedRender();
     video_.SetExternalEncoderFactory(encoder_factory);
     video_.SetExternalDecoderFactory(decoder_factory);
   }
@@ -66,7 +65,6 @@ class WebRtcMediaEngine2 :
                      WebRtcVideoDecoderFactory* decoder_factory) {
     voice_.SetAudioDeviceModule(adm, adm_sc);
     video_.SetVoiceEngine(&voice_);
-    video_.EnableTimedRender();
   }
 };
 #endif  // WEBRTC_CHROMIUM_BUILD
diff --git a/talk/media/webrtc/webrtcvideoengine.cc b/talk/media/webrtc/webrtcvideoengine.cc
index ed40911df6..0e13b9746b 100644
--- a/talk/media/webrtc/webrtcvideoengine.cc
+++ b/talk/media/webrtc/webrtcvideoengine.cc
@@ -81,6 +81,9 @@ const int kMaxVideoBitrate = 2000;
 
 const int kCpuMonitorPeriodMs = 2000;  // 2 seconds.
 
+// TODO(pthatcher): Figure out what the proper value here is, or if we
+// can just remove this altogether.
+static const int kDefaultRenderDelayMs = 100;
 
 static const int kDefaultLogSeverity = rtc::LS_WARNING;
 
@@ -951,6 +954,9 @@ void WebRtcVideoEngine::Construct(ViEWrapper* vie_wrapper,
     LOG(LS_ERROR) << "Failed to initialize list of supported codec types";
   }
 
+  // Consider jitter, packet loss, etc when rendering.  This will
+  // theoretically make rendering more smooth.
+  EnableTimedRender();
 
   // Load our RTP Header extensions.
   rtp_header_extensions_.push_back(
@@ -3392,6 +3398,11 @@ bool WebRtcVideoMediaChannel::ConfigureReceiving(int channel_id,
     return false;
   }
 
+  if (engine()->vie()->render()->SetExpectedRenderDelay(
+          channel_id, kDefaultRenderDelayMs)) {
+    LOG_RTCERR2(SetExpectedRenderDelay,
+                channel_id, kDefaultRenderDelayMs);
+  }
 
   if (engine_->vie()->rtp()->SetRembStatus(channel_id,
                                            kNotSending,