diff --git a/src/modules/rtp_rtcp/interface/rtp_rtcp.h b/src/modules/rtp_rtcp/interface/rtp_rtcp.h
index 6b55202dde..02939113a1 100644
--- a/src/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/src/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -377,7 +377,7 @@ public:
     */
     virtual WebRtc_Word32 SetRTPKeepaliveStatus(
         const bool enable,
-        const WebRtc_Word8 unknownPayloadType,
+        const int unknownPayloadType,
         const WebRtc_UWord16 deltaTransmitTimeMS) = 0;
 
     /*
@@ -391,7 +391,7 @@ public:
     */
     virtual WebRtc_Word32 RTPKeepaliveStatus(
         bool* enable,
-        WebRtc_Word8* unknownPayloadType,
+        int* unknownPayloadType,
         WebRtc_UWord16* deltaTransmitTimeMS) const = 0;
 
     /*
diff --git a/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index cc71e7bac0..d3ab74fb39 100644
--- a/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -77,7 +77,10 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_METHOD2(IncomingPacket,
       WebRtc_Word32(const WebRtc_UWord8* incomingPacket, const WebRtc_UWord16 packetLength));
   MOCK_METHOD4(IncomingAudioNTP,
-      WebRtc_Word32(const WebRtc_UWord32 audioReceivedNTPsecs, const WebRtc_UWord32 audioReceivedNTPfrac, const WebRtc_UWord32 audioRTCPArrivalTimeSecs, const WebRtc_UWord32 audioRTCPArrivalTimeFrac));
+      WebRtc_Word32(const WebRtc_UWord32 audioReceivedNTPsecs,
+                    const WebRtc_UWord32 audioReceivedNTPfrac,
+                    const WebRtc_UWord32 audioRTCPArrivalTimeSecs,
+                    const WebRtc_UWord32 audioRTCPArrivalTimeFrac));
   MOCK_METHOD0(InitSender,
       WebRtc_Word32());
   MOCK_METHOD1(RegisterSendTransport,
@@ -85,15 +88,20 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_METHOD1(SetMaxTransferUnit,
       WebRtc_Word32(const WebRtc_UWord16 size));
   MOCK_METHOD3(SetTransportOverhead,
-      WebRtc_Word32(const bool TCP, const bool IPV6, const WebRtc_UWord8 authenticationOverhead));
+      WebRtc_Word32(const bool TCP, const bool IPV6,
+                    const WebRtc_UWord8 authenticationOverhead));
   MOCK_CONST_METHOD0(MaxPayloadLength,
       WebRtc_UWord16());
   MOCK_CONST_METHOD0(MaxDataPayloadLength,
       WebRtc_UWord16());
   MOCK_METHOD3(SetRTPKeepaliveStatus,
-      WebRtc_Word32(const bool enable, const WebRtc_Word8 unknownPayloadType, const WebRtc_UWord16 deltaTransmitTimeMS));
+      WebRtc_Word32(const bool enable,
+                    const int unknownPayloadType,
+                    const WebRtc_UWord16 deltaTransmitTimeMS));
   MOCK_CONST_METHOD3(RTPKeepaliveStatus,
-      WebRtc_Word32(bool* enable, WebRtc_Word8* unknownPayloadType, WebRtc_UWord16* deltaTransmitTimeMS));
+      WebRtc_Word32(bool* enable,
+                    int* unknownPayloadType,
+                    WebRtc_UWord16* deltaTransmitTimeMS));
   MOCK_CONST_METHOD0(RTPKeepalive,
       bool());
   MOCK_METHOD1(RegisterSendPayload,
@@ -147,7 +155,13 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
       int(WebRtc_UWord32* available_bandwidth));
   MOCK_METHOD7(SendOutgoingData,
-      WebRtc_Word32(const FrameType frameType, const WebRtc_Word8 payloadType, const WebRtc_UWord32 timeStamp, const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtpVideoHdr));
+      WebRtc_Word32(const FrameType frameType,
+                    const WebRtc_Word8 payloadType,
+                    const WebRtc_UWord32 timeStamp,
+                    const WebRtc_UWord8* payloadData,
+                    const WebRtc_UWord32 payloadSize,
+                    const RTPFragmentationHeader* fragmentation,
+                    const RTPVideoHeader* rtpVideoHdr));
   MOCK_METHOD1(RegisterIncomingRTCPCallback,
       WebRtc_Word32(RtcpFeedback* incomingMessagesCallback));
   MOCK_CONST_METHOD0(RTCP,
@@ -164,7 +178,8 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_CONST_METHOD4(RemoteNTP,
       WebRtc_Word32(WebRtc_UWord32 *ReceivedNTPsecs, WebRtc_UWord32 *ReceivedNTPfrac, WebRtc_UWord32 *RTCPArrivalTimeSecs, WebRtc_UWord32 *RTCPArrivalTimeFrac));
   MOCK_METHOD2(AddMixedCNAME,
-      WebRtc_Word32(const WebRtc_UWord32 SSRC, const WebRtc_Word8 cName[RTCP_CNAME_SIZE]));
+      WebRtc_Word32(const WebRtc_UWord32 SSRC,
+                    const char cName[RTCP_CNAME_SIZE]));
   MOCK_METHOD1(RemoveMixedCNAME,
       WebRtc_Word32(const WebRtc_UWord32 SSRC));
   MOCK_CONST_METHOD5(RTT,
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index d25436d39e..6087cc36d4 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -870,7 +870,7 @@ bool ModuleRtpRtcpImpl::RTPKeepalive() const {
 
 WebRtc_Word32 ModuleRtpRtcpImpl::RTPKeepaliveStatus(
     bool* enable,
-    WebRtc_Word8* unknownPayloadType,
+    int* unknownPayloadType,
     WebRtc_UWord16* deltaTransmitTimeMS) const {
   WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RTPKeepaliveStatus()");
 
@@ -881,7 +881,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::RTPKeepaliveStatus(
 
 WebRtc_Word32 ModuleRtpRtcpImpl::SetRTPKeepaliveStatus(
   bool enable,
-  WebRtc_Word8 unknownPayloadType,
+  const int unknownPayloadType,
   WebRtc_UWord16 deltaTransmitTimeMS) {
   if (enable) {
     WEBRTC_TRACE(
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index e16bf71c2a..19858d58cc 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -148,13 +148,15 @@ public:
     */
     virtual WebRtc_Word32 InitSender();
 
-    virtual WebRtc_Word32 SetRTPKeepaliveStatus(const bool enable,
-                                              const WebRtc_Word8 unknownPayloadType,
-                                              const WebRtc_UWord16 deltaTransmitTimeMS);
+    virtual WebRtc_Word32 SetRTPKeepaliveStatus(
+        const bool enable,
+        const int unknownPayloadType,
+        const WebRtc_UWord16 deltaTransmitTimeMS);
 
-    virtual WebRtc_Word32 RTPKeepaliveStatus(bool* enable,
-                                           WebRtc_Word8* unknownPayloadType,
-                                           WebRtc_UWord16* deltaTransmitTimeMS) const;
+    virtual WebRtc_Word32 RTPKeepaliveStatus(
+        bool* enable,
+        int* unknownPayloadType,
+        WebRtc_UWord16* deltaTransmitTimeMS) const;
 
     virtual bool RTPKeepalive() const;
 
diff --git a/src/modules/rtp_rtcp/source/rtp_sender.cc b/src/modules/rtp_rtcp/source/rtp_sender.cc
index 374ce5320e..7b0755aa31 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender.cc
@@ -361,7 +361,7 @@ RTPSender::RTPKeepalive() const
 
 WebRtc_Word32
 RTPSender::RTPKeepaliveStatus(bool* enable,
-                              WebRtc_Word8* unknownPayloadType,
+                              int* unknownPayloadType,
                               WebRtc_UWord16* deltaTransmitTimeMS) const
 {
     CriticalSectionScoped cs(_sendCritsect);
@@ -382,7 +382,7 @@ RTPSender::RTPKeepaliveStatus(bool* enable,
 }
 
 WebRtc_Word32 RTPSender::EnableRTPKeepalive(
-    const WebRtc_Word8 unknownPayloadType,
+    const int unknownPayloadType,
     const WebRtc_UWord16 deltaTransmitTimeMS) {
   CriticalSectionScoped cs(_sendCritsect);
 
diff --git a/src/modules/rtp_rtcp/source/rtp_sender.h b/src/modules/rtp_rtcp/source/rtp_sender.h
index 00012e130c..1a2cb826f8 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.h
+++ b/src/modules/rtp_rtcp/source/rtp_sender.h
@@ -199,12 +199,12 @@ public:
     /*
     *    Keep alive
     */
-    WebRtc_Word32 EnableRTPKeepalive( const WebRtc_Word8 unknownPayloadType,
-                                    const WebRtc_UWord16 deltaTransmitTimeMS);
+    WebRtc_Word32 EnableRTPKeepalive( const int unknownPayloadType,
+                                      const WebRtc_UWord16 deltaTransmitTimeMS);
 
     WebRtc_Word32 RTPKeepaliveStatus(bool* enable,
-                                   WebRtc_Word8* unknownPayloadType,
-                                   WebRtc_UWord16* deltaTransmitTimeMS) const;
+                                     int* unknownPayloadType,
+                                     WebRtc_UWord16* deltaTransmitTimeMS) const;
 
     WebRtc_Word32 DisableRTPKeepalive();
 
diff --git a/src/video_engine/include/vie_rtp_rtcp.h b/src/video_engine/include/vie_rtp_rtcp.h
index 1397222a6e..f618fae326 100644
--- a/src/video_engine/include/vie_rtp_rtcp.h
+++ b/src/video_engine/include/vie_rtp_rtcp.h
@@ -263,7 +263,7 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
   virtual int SetRTPKeepAliveStatus(
       const int video_channel,
       bool enable,
-      const char unknown_payload_type,
+      const int unknown_payload_type,
       const unsigned int delta_transmit_time_seconds =
           KDefaultDeltaTransmitTimeSeconds) = 0;
 
@@ -271,7 +271,7 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
   virtual int GetRTPKeepAliveStatus(
       const int video_channel,
       bool& enabled,
-      char& unkown_payload_type,
+      int& unkown_payload_type,
       unsigned int& delta_transmit_time_seconds) const = 0;
 
   // This function enables capturing of RTP packets to a binary file on a
diff --git a/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc b/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
index d39f79ae8f..10c1304c35 100644
--- a/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
+++ b/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
@@ -661,10 +661,10 @@ void ViEAutoTest::ViERtpRtcpAPITest()
     // RTP Keepalive
     //
     {
-        char setPT = 123;
+        int setPT = 123;
         unsigned int setDeltaTime = 10;
         bool enabled = false;
-        char getPT = 0;
+        int getPT = 0;
         unsigned int getDeltaTime = 0;
         EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
             tbChannel.videoChannel, true, 119));
diff --git a/src/video_engine/vie_channel.cc b/src/video_engine/vie_channel.cc
index be1bc6d677..60de282bef 100644
--- a/src/video_engine/vie_channel.cc
+++ b/src/video_engine/vie_channel.cc
@@ -1058,7 +1058,7 @@ int ViEChannel::GetEstimatedReceiveBandwidth(
 
 WebRtc_Word32 ViEChannel::SetKeepAliveStatus(
     const bool enable,
-    const WebRtc_Word8 unknown_payload_type,
+    const int unknown_payload_type,
     const WebRtc_UWord16 delta_transmit_timeMS) {
   WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
                "%s", __FUNCTION__);
@@ -1109,7 +1109,7 @@ WebRtc_Word32 ViEChannel::SetKeepAliveStatus(
 
 WebRtc_Word32 ViEChannel::GetKeepAliveStatus(
     bool& enabled,
-    WebRtc_Word8& unknown_payload_type,
+    int& unknown_payload_type,
     WebRtc_UWord16& delta_transmit_time_ms) {
   WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
                __FUNCTION__);
diff --git a/src/video_engine/vie_channel.h b/src/video_engine/vie_channel.h
index 2d9bcd257c..349a04b1a2 100644
--- a/src/video_engine/vie_channel.h
+++ b/src/video_engine/vie_channel.h
@@ -164,10 +164,10 @@ class ViEChannel
                          WebRtc_UWord32& nackBitrateSent) const;
   int GetEstimatedReceiveBandwidth(WebRtc_UWord32* estimated_bandwidth) const;
   WebRtc_Word32 SetKeepAliveStatus(const bool enable,
-                                   const WebRtc_Word8 unknown_payload_type,
+                                   const int unknown_payload_type,
                                    const WebRtc_UWord16 delta_transmit_timeMS);
   WebRtc_Word32 GetKeepAliveStatus(bool& enable,
-                                   WebRtc_Word8& unknown_payload_type,
+                                   int& unknown_payload_type,
                                    WebRtc_UWord16& delta_transmit_timeMS);
   WebRtc_Word32 StartRTPDump(const char file_nameUTF8[1024],
                              RTPDirections direction);
diff --git a/src/video_engine/vie_rtp_rtcp_impl.cc b/src/video_engine/vie_rtp_rtcp_impl.cc
index 497481c687..0aaf8e4c39 100644
--- a/src/video_engine/vie_rtp_rtcp_impl.cc
+++ b/src/video_engine/vie_rtp_rtcp_impl.cc
@@ -770,7 +770,7 @@ int ViERTP_RTCPImpl::GetEstimatedReceiveBandwidth(
 int ViERTP_RTCPImpl::SetRTPKeepAliveStatus(
     const int video_channel,
     bool enable,
-    const char unknown_payload_type,
+    const int unknown_payload_type,
     const unsigned int delta_transmit_time_seconds) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
                ViEId(shared_data_->instance_id(), video_channel),
@@ -801,7 +801,7 @@ int ViERTP_RTCPImpl::SetRTPKeepAliveStatus(
 int ViERTP_RTCPImpl::GetRTPKeepAliveStatus(
     const int video_channel,
     bool& enabled,
-    char& unknown_payload_type,
+    int& unknown_payload_type,
     unsigned int& delta_transmit_time_seconds) const {
   WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
                ViEId(shared_data_->instance_id(), video_channel),
diff --git a/src/video_engine/vie_rtp_rtcp_impl.h b/src/video_engine/vie_rtp_rtcp_impl.h
index 72e813159d..386c96f70f 100644
--- a/src/video_engine/vie_rtp_rtcp_impl.h
+++ b/src/video_engine/vie_rtp_rtcp_impl.h
@@ -96,12 +96,12 @@ class ViERTP_RTCPImpl
   virtual int SetRTPKeepAliveStatus(
       const int video_channel,
       bool enable,
-      const char unknown_payload_type,
+      const int unknown_payload_type,
       const unsigned int delta_transmit_time_seconds);
   virtual int GetRTPKeepAliveStatus(
       const int video_channel,
       bool& enabled,
-      char& unkown_payload_type,
+      int& unkown_payload_type,
       unsigned int& delta_transmit_time_seconds) const;
   virtual int StartRTPDump(const int video_channel,
                            const char file_nameUTF8[1024],
diff --git a/src/voice_engine/main/interface/voe_rtp_rtcp.h b/src/voice_engine/main/interface/voe_rtp_rtcp.h
index e26d85f231..9f8609eba6 100644
--- a/src/voice_engine/main/interface/voe_rtp_rtcp.h
+++ b/src/voice_engine/main/interface/voe_rtp_rtcp.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 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
@@ -191,12 +191,12 @@ public:
     // This functionality can maintain an existing Network Address Translator
     // (NAT) mapping while regular RTP is no longer transmitted.
     virtual int SetRTPKeepaliveStatus(
-        int channel, bool enable, unsigned char unknownPayloadType,
+        int channel, bool enable, int unknownPayloadType,
         int deltaTransmitTimeSeconds = 15) = 0;
 
     // Gets the RTP keepalive mechanism status.
     virtual int GetRTPKeepaliveStatus(
-        int channel, bool& enabled, unsigned char& unknownPayloadType,
+        int channel, bool& enabled, int& unknownPayloadType,
         int& deltaTransmitTimeSeconds) = 0;
 
     // Enables capturing of RTP packets to a binary file on a specific
diff --git a/src/voice_engine/main/source/channel.cc b/src/voice_engine/main/source/channel.cc
index 2b0aef804c..fb24494beb 100644
--- a/src/voice_engine/main/source/channel.cc
+++ b/src/voice_engine/main/source/channel.cc
@@ -5646,7 +5646,7 @@ Channel::GetFECStatus(bool& enabled, int& redPayloadtype)
 
 int
 Channel::SetRTPKeepaliveStatus(bool enable,
-                               unsigned char unknownPayloadType,
+                               int unknownPayloadType,
                                int deltaTransmitTimeSeconds)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
@@ -5673,11 +5673,11 @@ Channel::SetRTPKeepaliveStatus(bool enable,
 
 int
 Channel::GetRTPKeepaliveStatus(bool& enabled,
-                               unsigned char& unknownPayloadType,
+                               int& unknownPayloadType,
                                int& deltaTransmitTimeSeconds)
 {
     bool onOff(false);
-    WebRtc_Word8 payloadType(0);
+    int payloadType(0);
     WebRtc_UWord16 deltaTransmitTimeMS(0);
     if (_rtpRtcpModule.RTPKeepaliveStatus(&onOff, &payloadType,
                                           &deltaTransmitTimeMS) != 0)
diff --git a/src/voice_engine/main/source/channel.h b/src/voice_engine/main/source/channel.h
index 2f5b66a370..e26413e610 100644
--- a/src/voice_engine/main/source/channel.h
+++ b/src/voice_engine/main/source/channel.h
@@ -343,9 +343,9 @@ public:
     int GetRTPStatistics(CallStatistics& stats);
     int SetFECStatus(bool enable, int redPayloadtype);
     int GetFECStatus(bool& enabled, int& redPayloadtype);
-    int SetRTPKeepaliveStatus(bool enable, unsigned char unknownPayloadType,
+    int SetRTPKeepaliveStatus(bool enable, int unknownPayloadType,
                               int deltaTransmitTimeSeconds);
-    int GetRTPKeepaliveStatus(bool& enabled, unsigned char& unknownPayloadType,
+    int GetRTPKeepaliveStatus(bool& enabled, int& unknownPayloadType,
                               int& deltaTransmitTimeSeconds);
     int StartRTPDump(const char fileNameUTF8[1024], RTPDirections direction);
     int StopRTPDump(RTPDirections direction);
diff --git a/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc b/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc
index cbc4d0d7e1..bc7a5c85da 100644
--- a/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc
+++ b/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 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
@@ -575,7 +575,7 @@ int VoERTP_RTCPImpl::GetFECStatus(int channel,
 
 int VoERTP_RTCPImpl::SetRTPKeepaliveStatus(int channel,
                                            bool enable,
-                                           unsigned char unknownPayloadType,
+                                           int unknownPayloadType,
                                            int deltaTransmitTimeSeconds)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1), 
@@ -603,7 +603,7 @@ int VoERTP_RTCPImpl::SetRTPKeepaliveStatus(int channel,
 
 int VoERTP_RTCPImpl::GetRTPKeepaliveStatus(int channel,
                                            bool& enabled,
-                                           unsigned char& unknownPayloadType,
+                                           int& unknownPayloadType,
                                            int& deltaTransmitTimeSeconds)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
diff --git a/src/voice_engine/main/source/voe_rtp_rtcp_impl.h b/src/voice_engine/main/source/voe_rtp_rtcp_impl.h
index 3cdf162c30..d3a840d02a 100644
--- a/src/voice_engine/main/source/voe_rtp_rtcp_impl.h
+++ b/src/voice_engine/main/source/voe_rtp_rtcp_impl.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 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
@@ -90,12 +90,12 @@ public:
     // RTP keepalive mechanism (maintains NAT mappings associated to RTP flows)
     virtual int SetRTPKeepaliveStatus(int channel,
                                       bool enable,
-                                      unsigned char unknownPayloadType,
+                                      int unknownPayloadType,
                                       int deltaTransmitTimeSeconds = 15);
 
     virtual int GetRTPKeepaliveStatus(int channel,
                                       bool& enabled,
-                                      unsigned char& unknownPayloadType,
+                                      int& unknownPayloadType,
                                       int& deltaTransmitTimeSeconds);
 
     // FEC
diff --git a/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc b/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc
index afd0820d17..666bb4b085 100644
--- a/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc
+++ b/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 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
@@ -43,7 +43,7 @@ TEST_F(RtpRtcpBeforeStreamingTest,
 }
 
 TEST_F(RtpRtcpBeforeStreamingTest, RtpKeepAliveStatusIsOffByDefault) {
-  unsigned char payload_type;
+  int payload_type;
   int delta_seconds;
   bool on;
 
@@ -56,7 +56,7 @@ TEST_F(RtpRtcpBeforeStreamingTest, RtpKeepAliveStatusIsOffByDefault) {
 }
 
 TEST_F(RtpRtcpBeforeStreamingTest, SetRtpKeepAliveDealsWithInvalidParameters) {
-  unsigned char payload_type;
+  int payload_type;
   int delta_seconds;
   bool on;
 
@@ -90,7 +90,7 @@ TEST_F(RtpRtcpBeforeStreamingTest,
   EXPECT_EQ(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
       channel_, true, 1));
 
-  unsigned char payload_type;
+  int payload_type;
   int delta_seconds;
   bool on;
 
diff --git a/src/voice_engine/main/test/auto_test/voe_extended_test.cc b/src/voice_engine/main/test/auto_test/voe_extended_test.cc
index 8b4ad455b2..11b41af874 100644
--- a/src/voice_engine/main/test/auto_test/voe_extended_test.cc
+++ b/src/voice_engine/main/test/auto_test/voe_extended_test.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 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
@@ -7149,7 +7149,7 @@ int VoEExtendedTest::TestRTP_RTCP() {
   ANL();
 
   TEST(GetRTPKeepaliveStatus);
-  unsigned char pt;
+  int pt;
   int dT;
   TEST_MUSTPASS(!rtp_rtcp->GetRTPKeepaliveStatus(-1, enabled, pt, dT));
   MARK();
@@ -7609,8 +7609,7 @@ int VoEExtendedTest::TestVolumeControl()
 
   TEST_MUSTPASS(voe_base_->Init());
   TEST_MUSTPASS(voe_base_->CreateChannel());
-#if (defined _TEST_HARDWARE_ && (!defined(MAC_IPHONE) && \
-    !defined(WEBRTC_ANDROID)))
+#if (defined _TEST_HARDWARE_ && (!defined(MAC_IPHONE)))
 #if defined(_WIN32)
   TEST_MUSTPASS(hardware->SetRecordingDevice(-1));
   TEST_MUSTPASS(hardware->SetPlayoutDevice(-1));