diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index a496316b42..016323892c 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -99,6 +99,9 @@ const int kMaxTelephoneEventCode = 255;
 const int kMinTelephoneEventDuration = 100;
 const int kMaxTelephoneEventDuration = 60000;   // Actual limit is 2^16
 
+const int kMinPayloadType = 0;
+const int kMaxPayloadType = 127;
+
 class ProxySink : public webrtc::AudioSinkInterface {
  public:
   ProxySink(AudioSinkInterface* sink) : sink_(sink) { RTC_DCHECK(sink); }
@@ -1549,12 +1552,16 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   // TODO(solenberg): Validate input - that payload types don't overlap, are
   //                  within range, filter out codecs we don't support,
-  //                  redundant codecs etc.
+  //                  redundant codecs etc - the same way it is done for
+  //                  RtpHeaderExtensions.
 
   // Find the DTMF telephone event "codec" payload type.
   dtmf_payload_type_ = rtc::Optional<int>();
   for (const AudioCodec& codec : codecs) {
     if (IsCodec(codec, kDtmfCodecName)) {
+      if (codec.id < kMinPayloadType || codec.id > kMaxPayloadType) {
+        return false;
+      }
       dtmf_payload_type_ = rtc::Optional<int>(codec.id);
       break;
     }
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 1fe9fc1ab6..4fd5918efd 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -1661,6 +1661,27 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) {
   EXPECT_TRUE(channel_->CanInsertDtmf());
 }
 
+// Test that payload type range is limited for telephone-event codec.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) {
+  EXPECT_TRUE(SetupEngineWithSendStream());
+  cricket::AudioSendParameters parameters;
+  parameters.codecs.push_back(kTelephoneEventCodec);
+  parameters.codecs.push_back(kIsacCodec);
+  parameters.codecs[0].id = 0;  // DTMF
+  parameters.codecs[1].id = 96;
+  EXPECT_TRUE(channel_->SetSendParameters(parameters));
+  EXPECT_TRUE(channel_->CanInsertDtmf());
+  parameters.codecs[0].id = 128;  // DTMF
+  EXPECT_FALSE(channel_->SetSendParameters(parameters));
+  EXPECT_FALSE(channel_->CanInsertDtmf());
+  parameters.codecs[0].id = 127;
+  EXPECT_TRUE(channel_->SetSendParameters(parameters));
+  EXPECT_TRUE(channel_->CanInsertDtmf());
+  parameters.codecs[0].id = -1;  // DTMF
+  EXPECT_FALSE(channel_->SetSendParameters(parameters));
+  EXPECT_FALSE(channel_->CanInsertDtmf());
+}
+
 // Test that we can set send codecs even with CN codec as the first
 // one on the list.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNOnTop) {
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 2c747af12b..8f1d0c18bd 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -361,7 +361,7 @@ void Channel::OnPlayTelephoneEvent(uint8_t event,
                " volume=%u)",
                event, lengthMs, volume);
 
-  if (!_playOutbandDtmfEvent || (event > 15)) {
+  if (!_playOutbandDtmfEvent || event > 15) {
     // Ignore callback since feedback is disabled or event is not a
     // Dtmf tone event.
     return;
@@ -2227,18 +2227,14 @@ int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
   return 0;
 }
 
-int Channel::SetSendTelephoneEventPayloadType(unsigned char type) {
+int Channel::SetSendTelephoneEventPayloadType(int payload_type) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetSendTelephoneEventPayloadType()");
-  if (type > 127) {
-    _engineStatisticsPtr->SetLastError(
-        VE_INVALID_ARGUMENT, kTraceError,
-        "SetSendTelephoneEventPayloadType() invalid type");
-    return -1;
-  }
-  CodecInst codec = {};
+  RTC_DCHECK_LE(0, payload_type);
+  RTC_DCHECK_GE(127, payload_type);
+  CodecInst codec = {0};
   codec.plfreq = 8000;
-  codec.pltype = type;
+  codec.pltype = payload_type;
   memcpy(codec.plname, "telephone-event", 16);
   if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
     _rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index f1f6ba2567..727a4e2edd 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -294,9 +294,9 @@ class Channel
   // VoEVideoSyncExtended
   int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
 
-  // VoEDtmf
+  // DTMF
   int SendTelephoneEventOutband(int event, int duration_ms);
-  int SetSendTelephoneEventPayloadType(unsigned char type);
+  int SetSendTelephoneEventPayloadType(int payload_type);
 
   // VoEAudioProcessingImpl
   int UpdateRxVadDetection(AudioFrame& audioFrame);