diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
index 3a0105d15f..7fcef2099c 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -733,10 +733,14 @@ class RtpRtcp : public Module {
 
     /*
     *   Turn negative acknowledgement requests on/off
+    *   |max_reordering_threshold| should be set to how much a retransmitted
+    *   packet can be expected to be reordered (in sequence numbers) compared to
+    *   a packet which has not been retransmitted.
     *
     *   return -1 on failure else 0
     */
-    virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method) = 0;
+    virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method,
+                                        int max_reordering_threshold) = 0;
 
     /*
      *  TODO(holmer): Propagate this API to VideoEngine.
@@ -774,7 +778,7 @@ class RtpRtcp : public Module {
     */
     virtual WebRtc_Word32 SetStorePacketsStatus(
         const bool enable,
-        const WebRtc_UWord16 numberToStore = 200) = 0;
+        const WebRtc_UWord16 numberToStore) = 0;
 
     /**************************************************************************
     *
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 52a9c5fba5..e4f3c1c83a 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -233,8 +233,8 @@ class MockRtpRtcp : public RtpRtcp {
       void(WebRtc_UWord16 bandWidthKbit));
   MOCK_CONST_METHOD0(NACK,
       NACKMethod());
-  MOCK_METHOD1(SetNACKStatus,
-      WebRtc_Word32(const NACKMethod method));
+  MOCK_METHOD2(SetNACKStatus,
+      WebRtc_Word32(const NACKMethod method, int oldestSequenceNumberToNack));
   MOCK_CONST_METHOD0(SelectiveRetransmissions,
       int());
   MOCK_METHOD1(SetSelectiveRetransmissions,
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index d05dd2d5cd..fba1818dd8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -800,7 +800,6 @@ RTCPReceiver::HandleNACK(RTCPUtility::RTCPParserV2& rtcpParser,
         rtcpParser.Iterate();
         return;
     }
-
     rtcpPacketInformation.ResetNACKPacketIdArray();
 
     RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
@@ -1271,13 +1270,11 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
     _rtpRtcp.OnRequestSendReport();
   }
   if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack) {
-    if (rtcpPacketInformation.nackSequenceNumbersLength > 0) {
+    if (rtcpPacketInformation.nackSequenceNumbers.size() > 0) {
       WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
                    "SIG [RTCP] Incoming NACK length:%d",
-                   rtcpPacketInformation.nackSequenceNumbersLength);
-      _rtpRtcp.OnReceivedNACK(
-          rtcpPacketInformation.nackSequenceNumbersLength,
-          rtcpPacketInformation.nackSequenceNumbers);
+                   rtcpPacketInformation.nackSequenceNumbers.size());
+      _rtpRtcp.OnReceivedNACK(rtcpPacketInformation.nackSequenceNumbers);
     }
   }
   {
@@ -1451,4 +1448,5 @@ void RTCPReceiver::PacketTimeout()
         _cbRtcpFeedback->OnRTCPPacketTimeout(_id);
     }
 }
+
 } // namespace webrtc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
index 09c8ba4ffd..4115f5d027 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc
@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "rtcp_receiver_help.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
 
 #include <string.h>  // memset
 #include <cassert>  // assert
 
-#include "modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
 using namespace RTCPHelp;
@@ -21,8 +21,7 @@ using namespace RTCPHelp;
 RTCPPacketInformation::RTCPPacketInformation()
     : rtcpPacketTypeFlags(0),
       remoteSSRC(0),
-      nackSequenceNumbers(0),
-      nackSequenceNumbersLength(0),
+      nackSequenceNumbers(),
       applicationSubType(0),
       applicationName(0),
       applicationData(),
@@ -44,7 +43,6 @@ RTCPPacketInformation::RTCPPacketInformation()
 
 RTCPPacketInformation::~RTCPPacketInformation()
 {
-    delete [] nackSequenceNumbers;
     delete [] applicationData;
     delete VoIPMetric;
 }
@@ -84,23 +82,16 @@ void RTCPPacketInformation::AddApplicationData(const WebRtc_UWord8* data,
 void
 RTCPPacketInformation::ResetNACKPacketIdArray()
 {
-    if (NULL == nackSequenceNumbers)
-    {
-        nackSequenceNumbers = new WebRtc_UWord16[NACK_PACKETS_MAX_SIZE];
-    }
-    nackSequenceNumbersLength = 0;
+  nackSequenceNumbers.clear();
 }
 
 void
 RTCPPacketInformation::AddNACKPacket(const WebRtc_UWord16 packetID)
 {
-    assert(nackSequenceNumbers);
-
-    WebRtc_UWord16& idx = nackSequenceNumbersLength;
-    if (idx < NACK_PACKETS_MAX_SIZE)
-    {
-        nackSequenceNumbers[idx++] = packetID;
-    }
+  if (nackSequenceNumbers.size() >= kSendSideNackListSizeSanity) {
+    return;
+  }
+  nackSequenceNumbers.push_back(packetID);
 }
 
 void
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
index 8af7b66949..1d538f0b48 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
@@ -11,12 +11,14 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
 
-#include <vector>
+#include <list>
 
-#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"  // RTCPReportBlock
-#include "modules/rtp_rtcp/source/rtcp_utility.h"
-#include "modules/rtp_rtcp/source/tmmbr_help.h"
-#include "typedefs.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"  // RTCPReportBlock
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
+#include "webrtc/system_wrappers/interface/constructor_magic.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 namespace RTCPHelp
@@ -44,8 +46,7 @@ public:
     WebRtc_UWord32  rtcpPacketTypeFlags; // RTCPPacketTypeFlags bit field
     WebRtc_UWord32  remoteSSRC;
 
-    WebRtc_UWord16* nackSequenceNumbers;
-    WebRtc_UWord16  nackSequenceNumbersLength;
+    std::list<uint16_t> nackSequenceNumbers;
 
     WebRtc_UWord8   applicationSubType;
     WebRtc_UWord32  applicationName;
@@ -69,6 +70,9 @@ public:
     uint32_t rtp_timestamp;
 
     RTCPVoIPMetric*  VoIPMetric;
+
+private:
+    DISALLOW_COPY_AND_ASSIGN(RTCPPacketInformation);
 };
 
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index 11d158accb..ad4b9090d6 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -205,7 +205,29 @@ class RtcpReceiverTest : public ::testing::Test {
     RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
     int result = rtcp_receiver_->IncomingRTCPPacket(rtcpPacketInformation,
                                                     &rtcpParser);
-    rtcp_packet_info_ = rtcpPacketInformation;
+    // The NACK list is on purpose not copied below as it isn't needed by the
+    // test.
+    rtcp_packet_info_.rtcpPacketTypeFlags =
+        rtcpPacketInformation.rtcpPacketTypeFlags;
+    rtcp_packet_info_.remoteSSRC = rtcpPacketInformation.remoteSSRC;
+    rtcp_packet_info_.applicationSubType =
+        rtcpPacketInformation.applicationSubType;
+    rtcp_packet_info_.applicationName = rtcpPacketInformation.applicationName;
+    rtcp_packet_info_.reportBlock = rtcpPacketInformation.reportBlock;
+    rtcp_packet_info_.fractionLost = rtcpPacketInformation.fractionLost;
+    rtcp_packet_info_.roundTripTime = rtcpPacketInformation.roundTripTime;
+    rtcp_packet_info_.lastReceivedExtendedHighSeqNum =
+        rtcpPacketInformation.lastReceivedExtendedHighSeqNum;
+    rtcp_packet_info_.jitter = rtcpPacketInformation.jitter;
+    rtcp_packet_info_.interArrivalJitter =
+        rtcpPacketInformation.interArrivalJitter;
+    rtcp_packet_info_.sliPictureId = rtcpPacketInformation.sliPictureId;
+    rtcp_packet_info_.rpsiPictureId = rtcpPacketInformation.rpsiPictureId;
+    rtcp_packet_info_.receiverEstimatedMaxBitrate =
+        rtcpPacketInformation.receiverEstimatedMaxBitrate;
+    rtcp_packet_info_.ntp_secs = rtcpPacketInformation.ntp_secs;
+    rtcp_packet_info_.ntp_frac = rtcpPacketInformation.ntp_frac;
+    rtcp_packet_info_.rtp_timestamp = rtcpPacketInformation.rtp_timestamp;
     return result;
   }
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index e3cadbd5f7..89f42792a8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -80,8 +80,28 @@ class TestTransport : public Transport,
     EXPECT_TRUE(rtcpParser.IsValid());
     RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
     EXPECT_EQ(0, rtcp_receiver_->IncomingRTCPPacket(rtcpPacketInformation,
-                                                   &rtcpParser));
-    rtcp_packet_info_ = rtcpPacketInformation;
+                                                    &rtcpParser));
+    rtcp_packet_info_.rtcpPacketTypeFlags =
+        rtcpPacketInformation.rtcpPacketTypeFlags;
+    rtcp_packet_info_.remoteSSRC = rtcpPacketInformation.remoteSSRC;
+    rtcp_packet_info_.applicationSubType =
+        rtcpPacketInformation.applicationSubType;
+    rtcp_packet_info_.applicationName = rtcpPacketInformation.applicationName;
+    rtcp_packet_info_.reportBlock = rtcpPacketInformation.reportBlock;
+    rtcp_packet_info_.fractionLost = rtcpPacketInformation.fractionLost;
+    rtcp_packet_info_.roundTripTime = rtcpPacketInformation.roundTripTime;
+    rtcp_packet_info_.lastReceivedExtendedHighSeqNum =
+        rtcpPacketInformation.lastReceivedExtendedHighSeqNum;
+    rtcp_packet_info_.jitter = rtcpPacketInformation.jitter;
+    rtcp_packet_info_.interArrivalJitter =
+        rtcpPacketInformation.interArrivalJitter;
+    rtcp_packet_info_.sliPictureId = rtcpPacketInformation.sliPictureId;
+    rtcp_packet_info_.rpsiPictureId = rtcpPacketInformation.rpsiPictureId;
+    rtcp_packet_info_.receiverEstimatedMaxBitrate =
+        rtcpPacketInformation.receiverEstimatedMaxBitrate;
+    rtcp_packet_info_.ntp_secs = rtcpPacketInformation.ntp_secs;
+    rtcp_packet_info_.ntp_frac = rtcpPacketInformation.ntp_frac;
+    rtcp_packet_info_.rtp_timestamp = rtcpPacketInformation.rtp_timestamp;
 
     return packet_len;
   }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
index 52b20fc982..0c82657028 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
@@ -91,6 +91,7 @@ RTPReceiver::RTPReceiver(const WebRtc_Word32 id,
       last_report_jitter_transmission_time_offset_(0),
 
       nack_method_(kNackOff),
+      max_reordering_threshold_(kDefaultMaxReorderingThreshold),
       rtx_(false),
       ssrc_rtx_(0) {
   assert(incoming_audio_messages_callback &&
@@ -267,8 +268,16 @@ NACKMethod RTPReceiver::NACK() const {
 }
 
 // Turn negative acknowledgment requests on/off.
-WebRtc_Word32 RTPReceiver::SetNACKStatus(const NACKMethod method) {
+WebRtc_Word32 RTPReceiver::SetNACKStatus(const NACKMethod method,
+                                         int max_reordering_threshold) {
   CriticalSectionScoped lock(critical_section_rtp_receiver_);
+  if (max_reordering_threshold < 0) {
+    return -1;
+  } else if (method == kNackRtcp) {
+    max_reordering_threshold_ = max_reordering_threshold;
+  } else {
+    max_reordering_threshold_ = kDefaultMaxReorderingThreshold;
+  }
   nack_method_ = method;
   return 0;
 }
@@ -572,7 +581,7 @@ bool RTPReceiver::InOrderPacket(const WebRtc_UWord16 sequence_number) const {
   if (received_seq_max_ >= sequence_number) {
     // Detect wrap-around.
     if (!(received_seq_max_ > 0xff00 && sequence_number < 0x0ff)) {
-      if (received_seq_max_ - NACK_PACKETS_MAX_SIZE > sequence_number) {
+      if (received_seq_max_ - max_reordering_threshold_ > sequence_number) {
         // We have a restart of the remote side.
       } else {
         // we received a retransmit of a packet we already have.
@@ -582,7 +591,7 @@ bool RTPReceiver::InOrderPacket(const WebRtc_UWord16 sequence_number) const {
   } else {
     // Detect wrap-around.
     if (sequence_number > 0xff00 && received_seq_max_ < 0x0ff) {
-      if (received_seq_max_ - NACK_PACKETS_MAX_SIZE > sequence_number) {
+      if (received_seq_max_ - max_reordering_threshold_ > sequence_number) {
         // We have a restart of the remote side
       } else {
         // We received a retransmit of a packet we already have
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver.h
index aac9e29842..854c556317 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver.h
@@ -88,7 +88,8 @@ class RTPReceiver : public Bitrate {
   NACKMethod NACK() const ;
 
   // Turn negative acknowledgement requests on/off.
-  WebRtc_Word32 SetNACKStatus(const NACKMethod method);
+  WebRtc_Word32 SetNACKStatus(const NACKMethod method,
+                              int max_reordering_threshold);
 
   // Returns the last received timestamp.
   virtual WebRtc_UWord32 TimeStamp() const;
@@ -159,7 +160,6 @@ class RTPReceiver : public Bitrate {
   virtual WebRtc_Word8 REDPayloadType() const;
 
   bool HaveNotReceivedPackets() const;
- protected:
 
   virtual bool RetransmitOfOldPacket(const WebRtc_UWord16 sequence_number,
                                      const WebRtc_UWord32 rtp_time_stamp) const;
@@ -184,7 +184,6 @@ class RTPReceiver : public Bitrate {
   void UpdateNACKBitRate(WebRtc_Word32 bytes, WebRtc_UWord32 now);
   bool ProcessNACKBitRate(WebRtc_UWord32 now);
 
- private:
   RTPPayloadRegistry*             rtp_payload_registry_;
   scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_;
 
@@ -243,6 +242,7 @@ class RTPReceiver : public Bitrate {
   mutable WebRtc_UWord32    last_report_jitter_transmission_time_offset_;
 
   NACKMethod nack_method_;
+  int max_reordering_threshold_;
 
   bool rtx_;
   WebRtc_UWord32 ssrc_rtx_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
index e128a10ffc..c8886365e3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
@@ -18,8 +18,10 @@ enum { kRtpRtcpMaxIdleTimeProcess = 5,
        kRtpRtcpPacketTimeoutProcessTimeMs = 100,
        kRtpRtcpRttProcessTimeMs = 1000 };
 
-enum { NACK_PACKETS_MAX_SIZE    = 256 }; // in packets
 enum { NACK_BYTECOUNT_SIZE      = 60};   // size of our NACK history
+// A sanity for the NACK list parsing at the send-side.
+enum { kSendSideNackListSizeSanity = 20000 };
+enum { kDefaultMaxReorderingThreshold = 50 };  // In sequence numbers.
 
 enum { RTCP_INTERVAL_VIDEO_MS       = 1000 };
 enum { RTCP_INTERVAL_AUDIO_MS       = 5000 };
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 3cb799482b..9408361966 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -1477,14 +1477,15 @@ NACKMethod ModuleRtpRtcpImpl::NACK() const {
 }
 
 // Turn negative acknowledgment requests on/off.
-WebRtc_Word32 ModuleRtpRtcpImpl::SetNACKStatus(NACKMethod method) {
+WebRtc_Word32 ModuleRtpRtcpImpl::SetNACKStatus(
+    NACKMethod method, int max_reordering_threshold) {
   WEBRTC_TRACE(kTraceModuleCall,
                kTraceRtpRtcp,
                id_,
                "SetNACKStatus(%u)", method);
 
   nack_method_ = method;
-  rtp_receiver_->SetNACKStatus(method);
+  rtp_receiver_->SetNACKStatus(method, max_reordering_threshold);
   return 0;
 }
 
@@ -1516,10 +1517,6 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SendNACK(const WebRtc_UWord16* nack_list,
                id_,
                "SendNACK(size:%u)", size);
 
-  if (size > NACK_PACKETS_MAX_SIZE) {
-    RequestKeyFrame();
-    return -1;
-  }
   WebRtc_UWord16 avg_rtt = 0;
   rtcp_receiver_.RTT(rtp_receiver_->SSRC(), NULL, &avg_rtt, NULL, NULL);
 
@@ -2006,18 +2003,14 @@ WebRtc_UWord32 ModuleRtpRtcpImpl::SendTimeOfSendReport(
 }
 
 void ModuleRtpRtcpImpl::OnReceivedNACK(
-    const WebRtc_UWord16 nack_sequence_numbers_length,
-    const WebRtc_UWord16* nack_sequence_numbers) {
+    const std::list<uint16_t>& nack_sequence_numbers) {
   if (!rtp_sender_.StorePackets() ||
-      nack_sequence_numbers == NULL ||
-      nack_sequence_numbers_length == 0) {
+      nack_sequence_numbers.size() == 0) {
     return;
   }
   WebRtc_UWord16 avg_rtt = 0;
   rtcp_receiver_.RTT(rtp_receiver_->SSRC(), NULL, &avg_rtt, NULL, NULL);
-  rtp_sender_.OnReceivedNACK(nack_sequence_numbers_length,
-                             nack_sequence_numbers,
-                             avg_rtt);
+  rtp_sender_.OnReceivedNACK(nack_sequence_numbers, avg_rtt);
 }
 
 WebRtc_Word32 ModuleRtpRtcpImpl::LastReceivedNTP(
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 666612bda3..49f7a28da9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -312,7 +312,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   virtual NACKMethod NACK() const;
 
   // Turn negative acknowledgment requests on/off.
-  virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method);
+  virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method,
+                                      int max_reordering_threshold);
 
   virtual int SelectiveRetransmissions() const;
 
@@ -325,7 +326,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   // Store the sent packets, needed to answer to a negative acknowledgment
   // requests.
   virtual WebRtc_Word32 SetStorePacketsStatus(
-      const bool enable, const WebRtc_UWord16 number_to_store = 200);
+      const bool enable, const WebRtc_UWord16 number_to_store);
 
   // (APP) Application specific data.
   virtual WebRtc_Word32 SetRTCPApplicationSpecificData(
@@ -447,8 +448,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   void OnReceivedReferencePictureSelectionIndication(
       const WebRtc_UWord64 picture_id);
 
-  void OnReceivedNACK(const WebRtc_UWord16 nack_sequence_numbers_length,
-                      const WebRtc_UWord16* nack_sequence_numbers);
+  void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers);
 
   void OnRequestSendReport();
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index ae67e323d4..991f12ad69 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -514,8 +514,7 @@ int RTPSender::SetSelectiveRetransmissions(uint8_t settings) {
 }
 
 void RTPSender::OnReceivedNACK(
-    const WebRtc_UWord16 nack_sequence_numbers_length,
-    const WebRtc_UWord16 *nack_sequence_numbers,
+    const std::list<uint16_t>& nack_sequence_numbers,
     const WebRtc_UWord16 avg_rtt) {
   const WebRtc_Word64 now = clock_->TimeInMilliseconds();
   WebRtc_UWord32 bytes_re_sent = 0;
@@ -528,9 +527,9 @@ void RTPSender::OnReceivedNACK(
     return;
   }
 
-  for (WebRtc_UWord16 i = 0; i < nack_sequence_numbers_length; ++i) {
-    const WebRtc_Word32 bytes_sent = ReSendPacket(nack_sequence_numbers[i],
-                                                  5 + avg_rtt);
+  for (std::list<uint16_t>::const_iterator it = nack_sequence_numbers.begin();
+      it != nack_sequence_numbers.end(); ++it) {
+    const WebRtc_Word32 bytes_sent = ReSendPacket(*it, 5 + avg_rtt);
     if (bytes_sent > 0) {
       bytes_re_sent += bytes_sent;
     } else if (bytes_sent == 0) {
@@ -541,7 +540,7 @@ void RTPSender::OnReceivedNACK(
       // Failed to send one Sequence number. Give up the rest in this nack.
       WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
                    "Failed resending RTP packet %d, Discard rest of packets",
-                   nack_sequence_numbers[i]);
+                   *it);
       break;
     }
     // Delay bandwidth estimate (RTT * BW).
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 33d91c7940..dc8cf3ad1b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -159,8 +159,7 @@ class RTPSender : public Bitrate, public RTPSenderInterface {
   // NACK.
   int SelectiveRetransmissions() const;
   int SetSelectiveRetransmissions(uint8_t settings);
-  void OnReceivedNACK(const WebRtc_UWord16 nack_sequence_numbers_length,
-                      const WebRtc_UWord16 *nack_sequence_numbers,
+  void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers,
                       const WebRtc_UWord16 avg_rtt);
 
   void SetStorePacketsStatus(const bool enable,
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
index 4c9f36efee..aa822195cd 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -109,6 +109,6 @@ TEST_F(RtpRtcpAPITest, RTCP) {
   EXPECT_FALSE(module->TMMBR());
 
   EXPECT_EQ(kNackOff, module->NACK());
-  EXPECT_EQ(0, module->SetNACKStatus(kNackRtcp));
+  EXPECT_EQ(0, module->SetNACKStatus(kNackRtcp, 450));
   EXPECT_EQ(kNackRtcp, module->NACK());
 }
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_nack.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
index b72d0b349d..16e4601486 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
@@ -116,8 +116,8 @@ class RtpRtcpNackTest : public ::testing::Test {
 
     EXPECT_EQ(0, video_module_->SetRTCPStatus(kRtcpCompound));
     EXPECT_EQ(0, video_module_->SetSSRC(kTestSsrc));
-    EXPECT_EQ(0, video_module_->SetNACKStatus(kNackRtcp));
-    EXPECT_EQ(0, video_module_->SetStorePacketsStatus(true));
+    EXPECT_EQ(0, video_module_->SetNACKStatus(kNackRtcp, 450));
+    EXPECT_EQ(0, video_module_->SetStorePacketsStatus(true, 600));
     EXPECT_EQ(0, video_module_->SetSendingStatus(true));
     EXPECT_EQ(0, video_module_->SetSequenceNumber(kTestSequenceNumber));
     EXPECT_EQ(0, video_module_->SetStartTimestamp(111111));
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
index b87904ae9e..cf181a66e1 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -47,8 +47,8 @@ class RtpRtcpVideoTest : public ::testing::Test {
 
     EXPECT_EQ(0, video_module_->SetRTCPStatus(kRtcpCompound));
     EXPECT_EQ(0, video_module_->SetSSRC(test_ssrc_));
-    EXPECT_EQ(0, video_module_->SetNACKStatus(kNackRtcp));
-    EXPECT_EQ(0, video_module_->SetStorePacketsStatus(true));
+    EXPECT_EQ(0, video_module_->SetNACKStatus(kNackRtcp, 450));
+    EXPECT_EQ(0, video_module_->SetStorePacketsStatus(true, 600));
     EXPECT_EQ(0, video_module_->SetSendingStatus(true));
 
     transport_->SetSendModule(video_module_);
diff --git a/webrtc/modules/video_coding/main/interface/video_coding.h b/webrtc/modules/video_coding/main/interface/video_coding.h
index b5adf35e52..77bd9cebd1 100644
--- a/webrtc/modules/video_coding/main/interface/video_coding.h
+++ b/webrtc/modules/video_coding/main/interface/video_coding.h
@@ -546,6 +546,13 @@ public:
     virtual int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
                                           DecodeErrors errorMode) = 0;
 
+    // Sets the maximum number of sequence numbers that we are allowed to NACK
+    // and the oldest sequence number that we will consider to NACK. If a
+    // sequence number older than |max_packet_age_to_nack| is missing
+    // a key frame will be requested.
+    virtual void SetNackSettings(size_t max_nack_list_size,
+                                 int max_packet_age_to_nack) = 0;
+
     // Enables recording of debugging information.
     virtual int StartDebugRecording(const char* file_name_utf8) = 0;
 
diff --git a/webrtc/modules/video_coding/main/interface/video_coding_defines.h b/webrtc/modules/video_coding/main/interface/video_coding_defines.h
index 273c625e56..f36c990cb0 100644
--- a/webrtc/modules/video_coding/main/interface/video_coding_defines.h
+++ b/webrtc/modules/video_coding/main/interface/video_coding_defines.h
@@ -41,10 +41,6 @@ namespace webrtc {
 #define VCM_VP8_PAYLOAD_TYPE       120
 #define VCM_I420_PAYLOAD_TYPE      124
 
-enum VCMNackProperties {
-  kNackHistoryLength = 450
-};
-
 enum VCMVideoProtection {
   kProtectionNack,                // Both send-side and receive-side
   kProtectionNackSender,          // Send-side only
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.cc b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
index 1a9e29ffde..d8aefe8bf5 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.cc
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
@@ -95,12 +95,14 @@ VCMJitterBuffer::VCMJitterBuffer(Clock* clock,
       nack_mode_(kNoNack),
       low_rtt_nack_threshold_ms_(-1),
       high_rtt_nack_threshold_ms_(-1),
+      nack_seq_nums_internal_(),
       nack_seq_nums_(),
       nack_seq_nums_length_(0),
+      max_nack_list_size_(0),
+      max_packet_age_to_nack_(0),
       waiting_for_key_frame_(false) {
   memset(frame_buffers_, 0, sizeof(frame_buffers_));
   memset(receive_statistics_, 0, sizeof(receive_statistics_));
-  memset(nack_seq_nums_internal_, -1, sizeof(nack_seq_nums_internal_));
 
   for (int i = 0; i < kStartNumberOfFrames; i++) {
     frame_buffers_[i] = new VCMFrameBuffer();
@@ -144,11 +146,17 @@ void VCMJitterBuffer::CopyFrom(const VCMJitterBuffer& rhs) {
     first_packet_ = rhs.first_packet_;
     last_decoded_state_ =  rhs.last_decoded_state_;
     num_not_decodable_packets_ = rhs.num_not_decodable_packets_;
+    assert(max_nack_list_size_ == rhs.max_nack_list_size_);
+    assert(max_packet_age_to_nack_ == rhs.max_packet_age_to_nack_);
     memcpy(receive_statistics_, rhs.receive_statistics_,
            sizeof(receive_statistics_));
-    memcpy(nack_seq_nums_internal_, rhs.nack_seq_nums_internal_,
-           sizeof(nack_seq_nums_internal_));
-    memcpy(nack_seq_nums_, rhs.nack_seq_nums_, sizeof(nack_seq_nums_));
+    nack_seq_nums_internal_.resize(rhs.nack_seq_nums_internal_.size());
+    std::copy(rhs.nack_seq_nums_internal_.begin(),
+              rhs.nack_seq_nums_internal_.end(),
+              nack_seq_nums_internal_.begin());
+    nack_seq_nums_.resize(rhs.nack_seq_nums_.size());
+    std::copy(rhs.nack_seq_nums_.begin(), rhs.nack_seq_nums_.end(),
+              nack_seq_nums_.begin());
     for (int i = 0; i < kMaxNumberOfFrames; i++) {
       if (frame_buffers_[i] != NULL) {
         delete frame_buffers_[i];
@@ -810,6 +818,20 @@ void VCMJitterBuffer::SetNackMode(VCMNackMode mode,
   }
 }
 
+void VCMJitterBuffer::SetNackSettings(size_t max_nack_list_size,
+                                      int max_packet_age_to_nack) {
+  CriticalSectionScoped cs(crit_sect_);
+  assert(max_packet_age_to_nack >= 0);
+  if (max_packet_age_to_nack <= 0) {
+    return;
+  }
+  max_nack_list_size_ = max_nack_list_size;
+  max_packet_age_to_nack_ = max_packet_age_to_nack;
+  nack_seq_nums_internal_.resize(max_packet_age_to_nack_);
+  std::fill(nack_seq_nums_internal_.begin(), nack_seq_nums_internal_.end(), -1);
+  nack_seq_nums_.resize(max_nack_list_size_);
+}
+
 VCMNackMode VCMJitterBuffer::nack_mode() const {
   CriticalSectionScoped cs(crit_sect_);
   return nack_mode_;
@@ -861,8 +883,8 @@ uint16_t* VCMJitterBuffer::CreateNackList(uint16_t* nack_list_size,
     number_of_seq_num = high_seq_num - low_seq_num;
   }
 
-  if (number_of_seq_num > kNackHistoryLength) {
-    // NACK list has grown too big, flush and try to restart.
+  if (number_of_seq_num > max_packet_age_to_nack_) {
+    // Some of the missing packets are too old.
     WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
                  VCMId(vcm_id_, receiver_id_),
                  "Nack list too large, try to find a key frame and restart "
@@ -872,14 +894,14 @@ uint16_t* VCMJitterBuffer::CreateNackList(uint16_t* nack_list_size,
     // This NACK size will trigger a key frame request.
     bool found_key_frame = false;
 
-    while (number_of_seq_num > kNackHistoryLength) {
+    while (number_of_seq_num > max_packet_age_to_nack_) {
       found_key_frame = RecycleFramesUntilKeyFrame();
 
       if (!found_key_frame) {
         break;
       }
 
-      // Check if we still have too many packets in the jitter buffer.
+      // Check if we are still missing too old sequence numbers.
       low_seq_num = -1;
       high_seq_num = -1;
       GetLowHighSequenceNumbers(&low_seq_num, &high_seq_num);
@@ -937,16 +959,15 @@ uint16_t* VCMJitterBuffer::CreateNackList(uint16_t* nack_list_size,
       // Reaching thus far means we are going to update the NACK list
       // When in hybrid mode, we use the soft NACKing feature.
       if (nack_mode_ == kNackHybrid) {
-        nack_seq_nums_index = (*it)->BuildSoftNackList(nack_seq_nums_internal_,
-                                                       number_of_seq_num,
-                                                       nack_seq_nums_index,
-                                                       rtt_ms_);
+        nack_seq_nums_index = (*it)->BuildSoftNackList(
+            &nack_seq_nums_internal_[0], number_of_seq_num,
+            nack_seq_nums_index, rtt_ms_);
       } else {
         // Used when the frame is being processed by the decoding thread
         // don't need to use that info in this loop.
-        nack_seq_nums_index = (*it)->BuildHardNackList(nack_seq_nums_internal_,
-                                                       number_of_seq_num,
-                                                       nack_seq_nums_index);
+        nack_seq_nums_index = (*it)->BuildHardNackList(
+            &nack_seq_nums_internal_[0], number_of_seq_num,
+            nack_seq_nums_index);
       }
     }
   }
@@ -979,6 +1000,26 @@ uint16_t* VCMJitterBuffer::CreateNackList(uint16_t* nack_list_size,
     *nack_list_size = empty_index;
   }
 
+  if (*nack_list_size > max_nack_list_size_) {
+    // Too many packets missing. Better to skip ahead to the next key frame or
+    // to request one.
+    bool found_key_frame = RecycleFramesUntilKeyFrame();
+    if (!found_key_frame) {
+      // Set the last decoded sequence number to current high.
+      // This is to not get a large nack list again right away.
+      last_decoded_state_.SetSeqNum(static_cast<uint16_t>(high_seq_num));
+      // Set to trigger key frame signal.
+      *nack_list_size = 0xffff;
+      *list_extended = true;
+      WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1,
+                   "\tNo key frame found, request one. nack_list_size: "
+                   "%u", *nack_list_size);
+    } else {
+      *nack_list_size = 0;
+    }
+    return NULL;
+  }
+
   if (*nack_list_size > nack_seq_nums_length_) {
     // Larger list: NACK list was extended since the last call.
     *list_extended = true;
@@ -1006,7 +1047,7 @@ uint16_t* VCMJitterBuffer::CreateNackList(uint16_t* nack_list_size,
 
   nack_seq_nums_length_ = *nack_list_size;
 
-  return nack_seq_nums_;
+  return &nack_seq_nums_[0];
 }
 
 int64_t VCMJitterBuffer::LastDecodedTimestamp() const {
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.h b/webrtc/modules/video_coding/main/source/jitter_buffer.h
index 88631e49a7..e0b7c47cb5 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.h
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.h
@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_JITTER_BUFFER_H_
 
 #include <list>
+#include <vector>
 
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
@@ -49,8 +50,10 @@ struct VCMJitterSample {
 
 class VCMJitterBuffer {
  public:
-  VCMJitterBuffer(Clock* clock, int vcm_id = -1, int receiver_id = -1,
-                  bool master = true);
+  VCMJitterBuffer(Clock* clock,
+                  int vcm_id,
+                  int receiver_id,
+                  bool master);
   virtual ~VCMJitterBuffer();
 
   // Makes |this| a deep copy of |rhs|.
@@ -144,6 +147,9 @@ class VCMJitterBuffer {
   void SetNackMode(VCMNackMode mode, int low_rtt_nack_threshold_ms,
                    int high_rtt_nack_threshold_ms);
 
+  void SetNackSettings(size_t max_nack_list_size,
+                       int max_packet_age_to_nack);
+
   // Returns the current NACK mode.
   VCMNackMode nack_mode() const;
 
@@ -259,9 +265,11 @@ class VCMJitterBuffer {
   int low_rtt_nack_threshold_ms_;
   int high_rtt_nack_threshold_ms_;
   // Holds the internal NACK list (the missing sequence numbers).
-  int32_t nack_seq_nums_internal_[kNackHistoryLength];
-  uint16_t nack_seq_nums_[kNackHistoryLength];
+  std::vector<int> nack_seq_nums_internal_;
+  std::vector<uint16_t> nack_seq_nums_;
   unsigned int nack_seq_nums_length_;
+  size_t max_nack_list_size_;
+  int max_packet_age_to_nack_;  // Measured in sequence numbers.
   bool waiting_for_key_frame_;
 
   DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc b/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
index 7264b5ea26..3d2ad0c140 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
@@ -146,10 +146,14 @@ class TestRunningJitterBuffer : public ::testing::Test {
 
   virtual void SetUp() {
     clock_.reset(new SimulatedClock(0));
-    jitter_buffer_ = new VCMJitterBuffer(clock_.get());
+    max_nack_list_size_ = 250;
+    oldest_packet_to_nack_ = 450;
+    jitter_buffer_ = new VCMJitterBuffer(clock_.get(), -1, -1, true);
     stream_generator = new StreamGenerator(0, 0,
                                            clock_->TimeInMilliseconds());
     jitter_buffer_->Start();
+    jitter_buffer_->SetNackSettings(max_nack_list_size_,
+                                    oldest_packet_to_nack_);
     memset(data_buffer_, 0, kDataBufferSize);
   }
 
@@ -223,6 +227,8 @@ class TestRunningJitterBuffer : public ::testing::Test {
   VCMJitterBuffer* jitter_buffer_;
   StreamGenerator* stream_generator;
   scoped_ptr<SimulatedClock> clock_;
+  size_t max_nack_list_size_;
+  int oldest_packet_to_nack_;
   uint8_t data_buffer_[kDataBufferSize];
 };
 
@@ -300,18 +306,19 @@ TEST_F(TestJitterBufferNack, TestEmptyPackets) {
   EXPECT_TRUE(DecodeCompleteFrame());
 }
 
-TEST_F(TestJitterBufferNack, TestNackListFull) {
+TEST_F(TestJitterBufferNack, TestNackTooOldPackets) {
   // Insert a key frame and decode it.
   InsertFrame(kVideoFrameKey);
   EXPECT_TRUE(DecodeCompleteFrame());
 
-  // Generate and drop |kNackHistoryLength| packets to fill the NACK list.
-  DropFrame(kNackHistoryLength);
+  // Drop one frame and insert |kNackHistoryLength| to trigger NACKing a too
+  // old packet.
+  DropFrame(1);
   // Insert a frame which should trigger a recycle until the next key frame.
-  InsertFrame(kVideoFrameDelta);
+  InsertFrames(oldest_packet_to_nack_, kVideoFrameDelta);
   EXPECT_FALSE(DecodeCompleteFrame());
 
-  uint16_t nack_list_length = kNackHistoryLength;
+  uint16_t nack_list_length = max_nack_list_size_;
   bool extended;
   uint16_t* nack_list = jitter_buffer_->CreateNackList(&nack_list_length,
                                                        &extended);
@@ -319,8 +326,47 @@ TEST_F(TestJitterBufferNack, TestNackListFull) {
   EXPECT_TRUE(nack_list_length == 0xffff && nack_list == NULL);
 
   InsertFrame(kVideoFrameDelta);
+  // Waiting for a key frame.
   EXPECT_FALSE(DecodeCompleteFrame());
   EXPECT_FALSE(DecodeFrame());
+
+  InsertFrame(kVideoFrameKey);
+  // The next complete continuous frame isn't a key frame, but we're waiting
+  // for one.
+  EXPECT_FALSE(DecodeCompleteFrame());
+  // Skipping ahead to the key frame.
+  EXPECT_TRUE(DecodeFrame());
+}
+
+TEST_F(TestJitterBufferNack, TestNackListFull) {
+  // Insert a key frame and decode it.
+  InsertFrame(kVideoFrameKey);
+  EXPECT_TRUE(DecodeCompleteFrame());
+
+  // Generate and drop |kNackHistoryLength| packets to fill the NACK list.
+  DropFrame(max_nack_list_size_);
+  // Insert a frame which should trigger a recycle until the next key frame.
+  InsertFrame(kVideoFrameDelta);
+  EXPECT_FALSE(DecodeCompleteFrame());
+
+  uint16_t nack_list_length = max_nack_list_size_;
+  bool extended;
+  uint16_t* nack_list = jitter_buffer_->CreateNackList(&nack_list_length,
+                                                       &extended);
+  // Verify that the jitter buffer requests a key frame.
+  EXPECT_TRUE(nack_list_length == 0xffff && nack_list == NULL);
+
+  InsertFrame(kVideoFrameDelta);
+  // Waiting for a key frame.
+  EXPECT_FALSE(DecodeCompleteFrame());
+  EXPECT_FALSE(DecodeFrame());
+
+  InsertFrame(kVideoFrameKey);
+  // The next complete continuous frame isn't a key frame, but we're waiting
+  // for one.
+  EXPECT_FALSE(DecodeCompleteFrame());
+  // Skipping ahead to the key frame.
+  EXPECT_TRUE(DecodeFrame());
 }
 
 TEST_F(TestJitterBufferNack, TestNackBeforeDecode) {
diff --git a/webrtc/modules/video_coding/main/source/receiver.cc b/webrtc/modules/video_coding/main/source/receiver.cc
index efb3ecdc19..fc5357f829 100644
--- a/webrtc/modules/video_coding/main/source/receiver.cc
+++ b/webrtc/modules/video_coding/main/source/receiver.cc
@@ -345,6 +345,12 @@ void VCMReceiver::SetNackMode(VCMNackMode nackMode) {
   }
 }
 
+void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
+                                  int max_packet_age_to_nack) {
+  jitter_buffer_.SetNackSettings(max_nack_list_size,
+                                 max_packet_age_to_nack);
+}
+
 VCMNackMode VCMReceiver::NackMode() const {
   CriticalSectionScoped cs(crit_sect_);
   return jitter_buffer_.nack_mode();
diff --git a/webrtc/modules/video_coding/main/source/receiver.h b/webrtc/modules/video_coding/main/source/receiver.h
index 9d18e247b4..492d616492 100644
--- a/webrtc/modules/video_coding/main/source/receiver.h
+++ b/webrtc/modules/video_coding/main/source/receiver.h
@@ -59,6 +59,8 @@ class VCMReceiver {
 
   // NACK.
   void SetNackMode(VCMNackMode nackMode);
+  void SetNackSettings(size_t max_nack_list_size,
+                       int max_packet_age_to_nack);
   VCMNackMode NackMode() const;
   VCMNackStatus NackList(uint16_t* nackList, uint16_t* size);
 
diff --git a/webrtc/modules/video_coding/main/source/video_coding_impl.cc b/webrtc/modules/video_coding/main/source/video_coding_impl.cc
index 11725ad6ca..25e6c5f603 100644
--- a/webrtc/modules/video_coding/main/source/video_coding_impl.cc
+++ b/webrtc/modules/video_coding/main/source/video_coding_impl.cc
@@ -68,6 +68,7 @@ _bitStreamBeforeDecoder(NULL),
 _frameFromFile(),
 _keyRequestMode(kKeyOnError),
 _scheduleKeyRequest(false),
+max_nack_list_size_(0),
 
 _sendCritSect(CriticalSectionWrapper::CreateCriticalSection()),
 _encoder(),
@@ -166,21 +167,27 @@ VideoCodingModuleImpl::Process()
     }
 
     // Packet retransmission requests
+    // TODO(holmer): Add API for changing Process interval and make sure it's
+    // disabled when NACK is off.
     if (_retransmissionTimer.TimeUntilProcess() == 0)
     {
         _retransmissionTimer.Processed();
         if (_packetRequestCallback != NULL)
         {
-            WebRtc_UWord16 nackList[kNackHistoryLength];
-            WebRtc_UWord16 length = kNackHistoryLength;
-            const WebRtc_Word32 ret = NackList(nackList, length);
+            WebRtc_UWord16 length;
+            {
+                CriticalSectionScoped cs(_receiveCritSect);
+                length = max_nack_list_size_;
+            }
+            std::vector<uint16_t> nackList(length);
+            const WebRtc_Word32 ret = NackList(&nackList[0], length);
             if (ret != VCM_OK && returnValue == VCM_OK)
             {
                 returnValue = ret;
             }
             if (length > 0)
             {
-                _packetRequestCallback->ResendPackets(nackList, length);
+                _packetRequestCallback->ResendPackets(&nackList[0], length);
             }
         }
     }
@@ -517,7 +524,6 @@ WebRtc_Word32
 VideoCodingModuleImpl::SetVideoProtection(VCMVideoProtection videoProtection,
                                           bool enable)
 {
-
     switch (videoProtection)
     {
 
@@ -1372,6 +1378,17 @@ int VideoCodingModuleImpl::SetReceiverRobustnessMode(
   return VCM_OK;
 }
 
+void VideoCodingModuleImpl::SetNackSettings(
+    size_t max_nack_list_size, int max_packet_age_to_nack) {
+  if (max_nack_list_size != 0) {
+    CriticalSectionScoped cs(_receiveCritSect);
+    max_nack_list_size_ = max_nack_list_size;
+  }
+  _receiver.SetNackSettings(max_nack_list_size, max_packet_age_to_nack);
+  _dualReceiver.SetNackSettings(max_nack_list_size,
+                                max_packet_age_to_nack);
+}
+
 int VideoCodingModuleImpl::StartDebugRecording(const char* file_name_utf8) {
   CriticalSectionScoped cs(_sendCritSect);
   _encoderInputFile = fopen(file_name_utf8, "wb");
diff --git a/webrtc/modules/video_coding/main/source/video_coding_impl.h b/webrtc/modules/video_coding/main/source/video_coding_impl.h
index e09872e287..24a1f83085 100644
--- a/webrtc/modules/video_coding/main/source/video_coding_impl.h
+++ b/webrtc/modules/video_coding/main/source/video_coding_impl.h
@@ -58,8 +58,7 @@ enum VCMKeyRequestMode
 class VideoCodingModuleImpl : public VideoCodingModule
 {
 public:
-    VideoCodingModuleImpl(const WebRtc_Word32 id,
-                          Clock* clock);
+    VideoCodingModuleImpl(const WebRtc_Word32 id, Clock* clock);
 
     virtual ~VideoCodingModuleImpl();
 
@@ -259,6 +258,10 @@ public:
     // Set the receiver robustness mode.
     virtual int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
                                           DecodeErrors errorMode);
+
+    virtual void SetNackSettings(size_t max_nack_list_size,
+                                 int max_packet_age_to_nack);
+
     // Enables recording of debugging information.
     virtual int StartDebugRecording(const char* file_name_utf8);
 
@@ -295,6 +298,7 @@ private:
     VCMFrameBuffer                      _frameFromFile;
     VCMKeyRequestMode                   _keyRequestMode;
     bool                                _scheduleKeyRequest;
+    size_t                              max_nack_list_size_;
 
     CriticalSectionWrapper*             _sendCritSect; // Critical section for send side
     VCMGenericEncoder*                  _encoder;
diff --git a/webrtc/modules/video_coding/main/source/video_coding_impl_unittest.cc b/webrtc/modules/video_coding/main/source/video_coding_impl_unittest.cc
index 8143fae47c..14878e5c58 100644
--- a/webrtc/modules/video_coding/main/source/video_coding_impl_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/video_coding_impl_unittest.cc
@@ -46,6 +46,9 @@ class TestVideoCodingModule : public ::testing::Test {
                                                false));
     EXPECT_EQ(0, vcm_->RegisterExternalDecoder(&decoder_, kUnusedPayloadType,
                                                true));
+    const size_t kMaxNackListSize = 250;
+    const int kMaxPacketAgeToNack = 450;
+    vcm_->SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack);
     memset(&settings_, 0, sizeof(settings_));
     EXPECT_EQ(0, vcm_->Codec(kVideoCodecVP8, &settings_));
     settings_.numberOfSimulcastStreams = kNumberOfStreams;
diff --git a/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc b/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc
index ccf37eb076..ca7672fe6e 100644
--- a/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc
@@ -37,6 +37,9 @@ class VCMRobustnessTest : public ::testing::Test {
     vcm_ = VideoCodingModule::Create(0, clock_.get());
     ASSERT_TRUE(vcm_ != NULL);
     ASSERT_EQ(0, vcm_->InitializeReceiver());
+    const size_t kMaxNackListSize = 250;
+    const int kMaxPacketAgeToNack = 450;
+    vcm_->SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack);
     ASSERT_EQ(0, vcm_->RegisterFrameTypeCallback(&frame_type_callback_));
     ASSERT_EQ(0, vcm_->RegisterPacketRequestCallback(&request_callback_));
     ASSERT_EQ(VCM_OK, vcm_->Codec(kVideoCodecVP8, &video_codec_));
diff --git a/webrtc/modules/video_coding/main/test/jitter_buffer_test.cc b/webrtc/modules/video_coding/main/test/jitter_buffer_test.cc
index 69c4c20bc8..538818357b 100644
--- a/webrtc/modules/video_coding/main/test/jitter_buffer_test.cc
+++ b/webrtc/modules/video_coding/main/test/jitter_buffer_test.cc
@@ -106,7 +106,7 @@ int JitterBufferTest(CmdArgs& args)
     WebRtc_UWord8 data[1500];
     VCMPacket packet(data, size, seqNum, timeStamp, true);
 
-    VCMJitterBuffer jb(clock);
+    VCMJitterBuffer jb(clock, -1, -1, true);
 
     seqNum = 1234;
     timeStamp = 123*90;
diff --git a/webrtc/modules/video_coding/main/test/mt_rx_tx_test.cc b/webrtc/modules/video_coding/main/test/mt_rx_tx_test.cc
index b99c31ad61..fd959dda0d 100644
--- a/webrtc/modules/video_coding/main/test/mt_rx_tx_test.cc
+++ b/webrtc/modules/video_coding/main/test/mt_rx_tx_test.cc
@@ -239,7 +239,7 @@ int MTRxTxTest(CmdArgs& args)
     FecProtectionParams delta_params = protectionCallback.DeltaFecParameters();
     FecProtectionParams key_params = protectionCallback.KeyFecParameters();
     rtp->SetFecParameters(&delta_params, &key_params);
-    rtp->SetNACKStatus(nackEnabled ? kNackRtcp : kNackOff);
+    rtp->SetNACKStatus(nackEnabled ? kNackRtcp : kNackOff, kMaxPacketAgeToNack);
 
     vcm->SetChannelParameters((WebRtc_UWord32) bitRate,
                               (WebRtc_UWord8) lossRate, rttMS);
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.cc b/webrtc/modules/video_coding/main/test/rtp_player.cc
index de062249b1..a52a1cdd9d 100644
--- a/webrtc/modules/video_coding/main/test/rtp_player.cc
+++ b/webrtc/modules/video_coding/main/test/rtp_player.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "rtp_player.h"
+#include "webrtc/modules/video_coding/main/test/rtp_player.h"
 
 #include <cstdlib>
 #ifdef WIN32
@@ -18,9 +18,10 @@
 #include <arpa/inet.h>
 #endif
 
-#include "../source/internal_defines.h"
 #include "gtest/gtest.h"
-#include "rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/video_coding/main/source/internal_defines.h"
+#include "webrtc/modules/video_coding/main/test/test_util.h"
 #include "webrtc/system_wrappers/interface/clock.h"
 
 using namespace webrtc;
@@ -191,7 +192,8 @@ WebRtc_Word32 RTPPlayer::Initialize(const PayloadTypeList* payloadList)
         _randVec[i] = rand();
     }
     _randVecPos = 0;
-    WebRtc_Word32 ret = _rtpModule->SetNACKStatus(kNackOff);
+    WebRtc_Word32 ret = _rtpModule->SetNACKStatus(kNackOff,
+                                                  kMaxPacketAgeToNack);
     if (ret < 0)
     {
         return -1;
diff --git a/webrtc/modules/video_coding/main/test/test_util.h b/webrtc/modules/video_coding/main/test/test_util.h
index c5b8545bb0..525f1e3e78 100644
--- a/webrtc/modules/video_coding/main/test/test_util.h
+++ b/webrtc/modules/video_coding/main/test/test_util.h
@@ -22,6 +22,9 @@
 #include "module_common_types.h"
 #include "testsupport/fileutils.h"
 
+enum { kMaxNackListSize = 250 };
+enum { kMaxPacketAgeToNack = 450 };
+
 // Class used for passing command line arguments to tests
 class CmdArgs
 {
diff --git a/webrtc/modules/video_coding/main/test/video_rtp_play.cc b/webrtc/modules/video_coding/main/test/video_rtp_play.cc
index bd22168778..cc9194939a 100644
--- a/webrtc/modules/video_coding/main/test/video_rtp_play.cc
+++ b/webrtc/modules/video_coding/main/test/video_rtp_play.cc
@@ -194,6 +194,7 @@ int RtpPlay(CmdArgs& args)
     vcm->SetVideoProtection(protectionMethod, protectionEnabled);
     vcm->SetRenderDelay(renderDelayMs);
     vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
+    vcm->SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack);
 
     ret = 0;
 
diff --git a/webrtc/modules/video_coding/main/test/video_rtp_play_mt.cc b/webrtc/modules/video_coding/main/test/video_rtp_play_mt.cc
index 911eb5c2b3..7e4e065aa1 100644
--- a/webrtc/modules/video_coding/main/test/video_rtp_play_mt.cc
+++ b/webrtc/modules/video_coding/main/test/video_rtp_play_mt.cc
@@ -84,8 +84,7 @@ int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTe
                 protection == kProtectionNack ||
                 kProtectionNackFEC));
     Clock* clock = Clock::GetRealTimeClock();
-    VideoCodingModule* vcm =
-            VideoCodingModule::Create(1, clock);
+    VideoCodingModule* vcm = VideoCodingModule::Create(1, clock);
     RtpDataCallback dataCallback(vcm);
     std::string rtpFilename;
     rtpFilename = args.inputFile;
@@ -227,6 +226,7 @@ int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTe
     vcm->SetVideoProtection(protection, protectionEnabled);
     vcm->SetRenderDelay(renderDelayMs);
     vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
+    vcm->SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack);
 
     EventWrapper& waitEvent = *EventWrapper::Create();
 
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index eeee2f2833..0c137095a6 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -123,6 +123,7 @@ ViEChannel::ViEChannel(WebRtc_Word32 channel_id,
 
   rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
   vie_receiver_.SetRtpRtcpModule(rtp_rtcp_.get());
+  vcm_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack);
 }
 
 WebRtc_Word32 ViEChannel::Init() {
@@ -150,7 +151,8 @@ WebRtc_Word32 ViEChannel::Init() {
                  "%s: RTP::SetRTCPStatus failure", __FUNCTION__);
   }
   if (paced_sender_) {
-    if (rtp_rtcp_->SetStorePacketsStatus(true, kNackHistorySize) != 0) {
+    if (rtp_rtcp_->SetStorePacketsStatus(true, kSendSidePacketHistorySize) !=
+        0) {
       WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
                    "%s:SetStorePacketsStatus failure", __FUNCTION__);
     }
@@ -293,10 +295,10 @@ WebRtc_Word32 ViEChannel::SetSendCodec(const VideoCodec& video_codec,
                      "%s: RTP::SetRTCPStatus failure", __FUNCTION__);
       }
       if (nack_method != kNackOff) {
-        rtp_rtcp->SetStorePacketsStatus(true, kNackHistorySize);
-        rtp_rtcp->SetNACKStatus(nack_method);
+        rtp_rtcp->SetStorePacketsStatus(true, kSendSidePacketHistorySize);
+        rtp_rtcp->SetNACKStatus(nack_method, kMaxPacketAgeToNack);
       } else if (paced_sender_) {
-        rtp_rtcp->SetStorePacketsStatus(true, kNackHistorySize);
+        rtp_rtcp->SetStorePacketsStatus(true, kSendSidePacketHistorySize);
       }
       if (fec_enabled) {
         rtp_rtcp->SetGenericFECStatus(fec_enabled, payload_type_red,
@@ -618,7 +620,7 @@ WebRtc_Word32 ViEChannel::ProcessNACKRequest(const bool enable) {
                    "%s: Could not enable NACK, RTPC not on ", __FUNCTION__);
       return -1;
     }
-    if (rtp_rtcp_->SetNACKStatus(nackMethod) != 0) {
+    if (rtp_rtcp_->SetNACKStatus(nackMethod, kMaxPacketAgeToNack) != 0) {
       WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
                    "%s: Could not set NACK method %d", __FUNCTION__,
                    nackMethod);
@@ -626,7 +628,7 @@ WebRtc_Word32 ViEChannel::ProcessNACKRequest(const bool enable) {
     }
     WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
                  "%s: Using NACK method %d", __FUNCTION__, nackMethod);
-    rtp_rtcp_->SetStorePacketsStatus(true, kNackHistorySize);
+    rtp_rtcp_->SetStorePacketsStatus(true, kSendSidePacketHistorySize);
 
     vcm_.RegisterPacketRequestCallback(this);
 
@@ -636,8 +638,8 @@ WebRtc_Word32 ViEChannel::ProcessNACKRequest(const bool enable) {
          it != simulcast_rtp_rtcp_.end();
          it++) {
       RtpRtcp* rtp_rtcp = *it;
-      rtp_rtcp->SetNACKStatus(nackMethod);
-      rtp_rtcp->SetStorePacketsStatus(true, kNackHistorySize);
+      rtp_rtcp->SetNACKStatus(nackMethod, kMaxPacketAgeToNack);
+      rtp_rtcp->SetStorePacketsStatus(true, kSendSidePacketHistorySize);
     }
   } else {
     CriticalSectionScoped cs(rtp_rtcp_cs_.get());
@@ -648,13 +650,13 @@ WebRtc_Word32 ViEChannel::ProcessNACKRequest(const bool enable) {
       if (paced_sender_ == NULL) {
         rtp_rtcp->SetStorePacketsStatus(false, 0);
       }
-      rtp_rtcp->SetNACKStatus(kNackOff);
+      rtp_rtcp->SetNACKStatus(kNackOff, kMaxPacketAgeToNack);
     }
     vcm_.RegisterPacketRequestCallback(NULL);
     if (paced_sender_ == NULL) {
       rtp_rtcp_->SetStorePacketsStatus(false, 0);
     }
-    if (rtp_rtcp_->SetNACKStatus(kNackOff) != 0) {
+    if (rtp_rtcp_->SetNACKStatus(kNackOff, kMaxPacketAgeToNack) != 0) {
       WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
                    "%s: Could not turn off NACK", __FUNCTION__);
       return -1;
diff --git a/webrtc/video_engine/vie_defines.h b/webrtc/video_engine/vie_defines.h
index 42c91b5616..7495fcf394 100644
--- a/webrtc/video_engine/vie_defines.h
+++ b/webrtc/video_engine/vie_defines.h
@@ -75,7 +75,11 @@ enum { kViEMinRenderTimeoutTimeMs = 33 };
 enum { kViEDefaultRenderDelayMs = 10 };
 
 // ViERTP_RTCP
-enum { kNackHistorySize = 400 };
+enum { kSendSidePacketHistorySize = 600 };
+
+// NACK
+enum { kMaxPacketAgeToNack = 450 };  // In sequence numbers.
+enum { kMaxNackListSize = 250 };
 
 // Id definitions
 enum {