diff --git a/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc b/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
index 7271003044..08a4031922 100644
--- a/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
+++ b/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
@@ -686,6 +686,18 @@ void ViEAutoTest::ViERtpRtcpAPITest()
         tbChannel.videoChannel, false));
 
     // Buffering mode - sender side.
+    // Set VoE (required to set up stream-sync).
+    webrtc::VoiceEngine* voice_engine = webrtc::VoiceEngine::Create();
+    EXPECT_TRUE(NULL != voice_engine);
+    webrtc::VoEBase* voe_base = webrtc::VoEBase::GetInterface(voice_engine);
+    EXPECT_TRUE(NULL != voe_base);
+    EXPECT_EQ(0, voe_base->Init());
+    int audio_channel = voe_base->CreateChannel();
+    EXPECT_NE(-1, audio_channel);
+    EXPECT_EQ(0, ViE.base->SetVoiceEngine(voice_engine));
+    EXPECT_EQ(0, ViE.base->ConnectAudioChannel(tbChannel.videoChannel,
+                                               audio_channel));
+
     EXPECT_EQ(-1, ViE.rtp_rtcp->SetSenderBufferingMode(
         invalid_channel_id, 0));
     int invalid_delay = -1;
@@ -713,6 +725,12 @@ void ViEAutoTest::ViERtpRtcpAPITest()
     EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiverBufferingMode(
         tbChannel.videoChannel, 0));
 
+    EXPECT_EQ(0, ViE.base->DisconnectAudioChannel(tbChannel.videoChannel));
+    EXPECT_EQ(0, ViE.base->SetVoiceEngine(NULL));
+    EXPECT_EQ(0, voe_base->DeleteChannel(audio_channel));
+    voe_base->Release();
+    EXPECT_TRUE(webrtc::VoiceEngine::Delete(voice_engine));
+
     //***************************************************************
     //  Testing finished. Tear down Video Engine
     //***************************************************************