diff --git a/webrtc/video/payload_router.cc b/webrtc/video/payload_router.cc
index 8b29e818f0..f2f430904e 100644
--- a/webrtc/video/payload_router.cc
+++ b/webrtc/video/payload_router.cc
@@ -98,11 +98,6 @@ PayloadRouter::PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
 
 PayloadRouter::~PayloadRouter() {}
 
-size_t PayloadRouter::DefaultMaxPayloadLength() {
-  const size_t kIpUdpSrtpLength = 44;
-  return IP_PACKET_SIZE - kIpUdpSrtpLength;
-}
-
 void PayloadRouter::SetActive(bool active) {
   rtc::CritScope lock(&crit_);
   if (active_ == active)
@@ -149,17 +144,6 @@ EncodedImageCallback::Result PayloadRouter::OnEncodedImage(
   return Result(Result::OK, frame_id);
 }
 
-size_t PayloadRouter::MaxPayloadLength() const {
-  size_t min_payload_length = DefaultMaxPayloadLength();
-  rtc::CritScope lock(&crit_);
-  for (size_t i = 0; i < rtp_modules_.size(); ++i) {
-    size_t module_payload_length = rtp_modules_[i]->MaxDataPayloadLength();
-    if (module_payload_length < min_payload_length)
-      min_payload_length = module_payload_length;
-  }
-  return min_payload_length;
-}
-
 void PayloadRouter::OnBitrateAllocationUpdated(
     const BitrateAllocation& bitrate) {
   rtc::CritScope lock(&crit_);
diff --git a/webrtc/video/payload_router.h b/webrtc/video/payload_router.h
index 3b60ce23c0..f2d138dc2c 100644
--- a/webrtc/video/payload_router.h
+++ b/webrtc/video/payload_router.h
@@ -36,8 +36,6 @@ class PayloadRouter : public EncodedImageCallback {
                 int payload_type);
   ~PayloadRouter();
 
-  static size_t DefaultMaxPayloadLength();
-
   // PayloadRouter will only route packets if being active, all packets will be
   // dropped otherwise.
   void SetActive(bool active);
@@ -50,10 +48,6 @@ class PayloadRouter : public EncodedImageCallback {
       const CodecSpecificInfo* codec_specific_info,
       const RTPFragmentationHeader* fragmentation) override;
 
-  // Returns the maximum allowed data payload length, given the configured MTU
-  // and RTP headers.
-  size_t MaxPayloadLength() const;
-
   void OnBitrateAllocationUpdated(const BitrateAllocation& bitrate);
 
  private:
diff --git a/webrtc/video/payload_router_unittest.cc b/webrtc/video/payload_router_unittest.cc
index 935de5f3e9..e316695338 100644
--- a/webrtc/video/payload_router_unittest.cc
+++ b/webrtc/video/payload_router_unittest.cc
@@ -149,40 +149,6 @@ TEST(PayloadRouterTest, SendSimulcast) {
                 .error);
 }
 
-TEST(PayloadRouterTest, MaxPayloadLength) {
-  // Without any limitations from the modules, verify we get the max payload
-  // length for IP/UDP/SRTP with a MTU of 150 bytes.
-  const size_t kDefaultMaxLength = 1500 - 20 - 8 - 12 - 4;
-  NiceMock<MockRtpRtcp> rtp_1;
-  NiceMock<MockRtpRtcp> rtp_2;
-  std::vector<RtpRtcp*> modules;
-  modules.push_back(&rtp_1);
-  modules.push_back(&rtp_2);
-  PayloadRouter payload_router(modules, 42);
-
-  EXPECT_EQ(kDefaultMaxLength, PayloadRouter::DefaultMaxPayloadLength());
-  std::vector<VideoStream> streams(2);
-
-  // Modules return a higher length than the default value.
-  EXPECT_CALL(rtp_1, MaxDataPayloadLength())
-      .Times(1)
-      .WillOnce(Return(kDefaultMaxLength + 10));
-  EXPECT_CALL(rtp_2, MaxDataPayloadLength())
-      .Times(1)
-      .WillOnce(Return(kDefaultMaxLength + 10));
-  EXPECT_EQ(kDefaultMaxLength, payload_router.MaxPayloadLength());
-
-  // The modules return a value lower than default.
-  const size_t kTestMinPayloadLength = 1001;
-  EXPECT_CALL(rtp_1, MaxDataPayloadLength())
-      .Times(1)
-      .WillOnce(Return(kTestMinPayloadLength + 10));
-  EXPECT_CALL(rtp_2, MaxDataPayloadLength())
-      .Times(1)
-      .WillOnce(Return(kTestMinPayloadLength));
-  EXPECT_EQ(kTestMinPayloadLength, payload_router.MaxPayloadLength());
-}
-
 TEST(PayloadRouterTest, SimulcastTargetBitrate) {
   NiceMock<MockRtpRtcp> rtp_1;
   NiceMock<MockRtpRtcp> rtp_2;