diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
index dee05fe428..db56d789ec 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -304,6 +304,7 @@ class RTPPayloadAudioStrategy : public RTPPayloadStrategy {
     ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload;
     payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
     strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+    assert(frequency >= 1000);
     payload->typeSpecific.Audio.frequency = frequency;
     payload->typeSpecific.Audio.channels = channels;
     payload->typeSpecific.Audio.rate = rate;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index a1b44d8c46..4c4a76014a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -334,6 +334,8 @@ bool RtpReceiverImpl::RetransmitOfOldPacket(const RTPHeader& header,
 
   CriticalSectionScoped cs(critical_section_rtp_receiver_);
   uint32_t frequency_khz = header.payload_type_frequency / 1000;
+  assert(frequency_khz > 0);
+
   int64_t time_diff_ms = clock_->TimeInMilliseconds() -
       last_receive_time_;
 
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index b6b8c292f1..fdd2f6ce4c 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -2182,6 +2182,8 @@ int32_t Channel::ReceivedRTPPacket(const int8_t* data, int32_t length) {
                  "IncomingPacket invalid RTP header");
     return -1;
   }
+  header.payload_type_frequency =
+      rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType);
   bool retransmitted = IsPacketRetransmitted(header);
   bool in_order = rtp_receiver_->InOrderPacket(header.sequenceNumber);
   rtp_receive_statistics_->IncomingPacket(header, static_cast<uint16_t>(length),