diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index b0f50755dd..94853d2f6b 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -430,10 +430,11 @@ class RtpRtcp : public Module {
   // Audio
   // **************************************************************************
 
-  // Sets audio packet size, used to determine when it's time to send a DTMF
-  // packet in silence (CNG).
+  // This function is deprecated. It was previously used to determine when it
+  // was time to send a DTMF packet in silence (CNG).
   // Returns -1 on failure else 0.
-  virtual int32_t SetAudioPacketSize(uint16_t packet_size_samples) = 0;
+  RTC_DEPRECATED virtual int32_t SetAudioPacketSize(
+      uint16_t packet_size_samples) = 0;
 
   // Sends a TelephoneEvent tone using RFC 2833 (4733).
   // Returns -1 on failure else 0.
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index b80c8daf8b..b7b22d900b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -777,11 +777,9 @@ int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(
   return rtp_sender_.SendTelephoneEvent(key, time_ms, level);
 }
 
-// Set audio packet size, used to determine when it's time to send a DTMF
-// packet in silence (CNG).
 int32_t ModuleRtpRtcpImpl::SetAudioPacketSize(
     const uint16_t packet_size_samples) {
-  return rtp_sender_.SetAudioPacketSize(packet_size_samples);
+  return audio_ ? 0 : -1;
 }
 
 int32_t ModuleRtpRtcpImpl::SetAudioLevel(
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index bd978d05b5..9363a4716a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -260,8 +260,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
 
   // Audio part.
 
-  // Set audio packet size, used to determine when it's time to send a DTMF
-  // packet in silence (CNG).
+  // This function is deprecated. It was previously used to determine when it
+  // was time to send a DTMF packet in silence (CNG).
   int32_t SetAudioPacketSize(uint16_t packet_size_samples) override;
 
   // Send a TelephoneEvent tone using RFC 2833 (4733).
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 1b6df005d6..6dcff3ab37 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -1153,7 +1153,7 @@ int32_t RTPSender::SetAudioPacketSize(uint16_t packet_size_samples) {
   if (!audio_configured_) {
     return -1;
   }
-  return audio_->SetAudioPacketSize(packet_size_samples);
+  return 0;
 }
 
 int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 66db7502e9..080165a0bd 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -179,9 +179,9 @@ class RTPSender {
   // Send a DTMF tone using RFC 2833 (4733).
   int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
-  // Set audio packet size, used to determine when it's time to send a DTMF
-  // packet in silence (CNG).
-  int32_t SetAudioPacketSize(uint16_t packet_size_samples);
+  // This function is deprecated. It was previously used to determine when it
+  // was time to send a DTMF packet in silence (CNG).
+  RTC_DEPRECATED int32_t SetAudioPacketSize(uint16_t packet_size_samples);
 
   // Store the audio level in d_bov for
   // header-extension-for-audio-level-indication.
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 69eb208f10..0f32d4e786 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -31,14 +31,6 @@ RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
 
 RTPSenderAudio::~RTPSenderAudio() {}
 
-// set audio packet size, used to determine when it's time to send a DTMF packet
-// in silence (CNG)
-int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packet_size_samples) {
-  rtc::CritScope cs(&send_audio_critsect_);
-  packet_size_samples_ = packet_size_samples;
-  return 0;
-}
-
 int32_t RTPSenderAudio::RegisterAudioPayload(
     const char payloadName[RTP_PAYLOAD_NAME_SIZE],
     const int8_t payload_type,
@@ -132,19 +124,24 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
                                const uint8_t* payload_data,
                                size_t payload_size,
                                const RTPFragmentationHeader* fragmentation) {
+  // From RFC 4733:
+  // A source has wide latitude as to how often it sends event updates. A
+  // natural interval is the spacing between non-event audio packets. [...]
+  // Alternatively, a source MAY decide to use a different spacing for event
+  // updates, with a value of 50 ms RECOMMENDED.
+  constexpr int kDtmfIntervalTimeMs = 50;
   uint8_t audio_level_dbov = 0;
-  uint16_t packet_size_samples = 0;
   uint32_t dtmf_payload_freq = 0;
   {
     rtc::CritScope cs(&send_audio_critsect_);
     audio_level_dbov = audio_level_dbov_;
-    packet_size_samples = packet_size_samples_;
     dtmf_payload_freq = dtmf_payload_freq_;
   }
 
   // Check if we have pending DTMFs to send
   if (!dtmf_event_is_on_ && dtmf_queue_.PendingDtmf()) {
-    if ((clock_->TimeInMilliseconds() - dtmf_time_last_sent_) > 50) {
+    if ((clock_->TimeInMilliseconds() - dtmf_time_last_sent_) >
+        kDtmfIntervalTimeMs) {
       // New tone to play
       dtmf_timestamp_ = rtp_timestamp;
       if (dtmf_queue_.NextDtmf(&dtmf_current_event_)) {
@@ -163,7 +160,10 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
       // kEmptyFrame is used to drive the DTMF when in CN mode
       // it can be triggered more frequently than we want to send the
       // DTMF packets.
-      if (packet_size_samples > (rtp_timestamp - dtmf_timestamp_last_sent_)) {
+      const unsigned int dtmf_interval_time_rtp =
+          dtmf_payload_freq * kDtmfIntervalTimeMs / 1000;
+      if ((rtp_timestamp - dtmf_timestamp_last_sent_) <
+          dtmf_interval_time_rtp) {
         // not time to send yet
         return true;
       }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index ee4831c634..cf79120bb9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -42,10 +42,6 @@ class RTPSenderAudio {
                  size_t payload_size,
                  const RTPFragmentationHeader* fragmentation);
 
-  // set audio packet size, used to determine when it's time to send a DTMF
-  // packet in silence (CNG)
-  int32_t SetAudioPacketSize(uint16_t packet_size_samples);
-
   // Store the audio level in dBov for
   // header-extension-for-audio-level-indication.
   // Valid range is [0,100]. Actual value is negative.
@@ -69,8 +65,6 @@ class RTPSenderAudio {
 
   rtc::CriticalSection send_audio_critsect_;
 
-  uint16_t packet_size_samples_ GUARDED_BY(send_audio_critsect_) = 160;
-
   // DTMF.
   bool dtmf_event_is_on_ = false;
   bool dtmf_event_first_packet_sent_ = false;
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 6e2e91c0a1..8d95c7bfbb 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -1323,12 +1323,6 @@ int32_t Channel::SetSendCodec(const CodecInst& codec) {
     }
   }
 
-  if (_rtpRtcpModule->SetAudioPacketSize(codec.pacsize) != 0) {
-    WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "SetSendCodec() failed to set audio packet size");
-    return -1;
-  }
-
   return 0;
 }