diff --git a/api/data_channel_interface.h b/api/data_channel_interface.h
index a02bc5e..4f74918 100644
--- a/api/data_channel_interface.h
+++ b/api/data_channel_interface.h
@@ -183,12 +183,10 @@
 
   // Sends `data` to the remote peer. If the data can't be sent at the SCTP
   // level (due to congestion control), it's buffered at the data channel level,
-  // up to a maximum of MaxSendQueueSize(). If Send is called while this buffer
-  // is full, the data channel will be closed abruptly.
-  //
-  // So, it's important to use buffered_amount() and OnBufferedAmountChange to
-  // ensure the data channel is used efficiently but without filling this
-  // buffer.
+  // up to a maximum of MaxSendQueueSize().
+  // Returns false if the data channel is not in open state or if the send
+  // buffer is full.
+  // TODO(webrtc:13289): Return an RTCError with information about the failure.
   virtual bool Send(const DataBuffer& buffer) = 0;
 
   // Amount of bytes that can be queued for sending on the data channel.
diff --git a/pc/data_channel_unittest.cc b/pc/data_channel_unittest.cc
index 770892c..44c080b 100644
--- a/pc/data_channel_unittest.cc
+++ b/pc/data_channel_unittest.cc
@@ -544,23 +544,29 @@
   EXPECT_EQ(webrtc::InternalDataChannelInit::kNone, init2.open_handshake_role);
 }
 
-// Tests that the DataChannel is closed if the sending buffer is full.
-TEST_F(SctpDataChannelTest, ClosedWhenSendBufferFull) {
+// Tests that that Send() returns false if the sending buffer is full
+// and the channel stays open.
+TEST_F(SctpDataChannelTest, OpenWhenSendBufferFull) {
   SetChannelReady();
 
-  rtc::CopyOnWriteBuffer buffer(1024);
+  const size_t packetSize = 1024;
+
+  rtc::CopyOnWriteBuffer buffer(packetSize);
   memset(buffer.MutableData(), 0, buffer.size());
 
   webrtc::DataBuffer packet(buffer, true);
   provider_->set_send_blocked(true);
 
-  for (size_t i = 0; i < 16 * 1024 + 1; ++i) {
+  for (size_t i = 0;
+       i < webrtc::DataChannelInterface::MaxSendQueueSize() / packetSize; ++i) {
     EXPECT_TRUE(webrtc_data_channel_->Send(packet));
   }
 
-  EXPECT_TRUE(
-      webrtc::DataChannelInterface::kClosed == webrtc_data_channel_->state() ||
-      webrtc::DataChannelInterface::kClosing == webrtc_data_channel_->state());
+  // The sending buffer shoul be full, send returns false.
+  EXPECT_FALSE(webrtc_data_channel_->Send(packet));
+
+  EXPECT_TRUE(webrtc::DataChannelInterface::kOpen ==
+              webrtc_data_channel_->state());
 }
 
 // Tests that the DataChannel is closed on transport errors.
diff --git a/pc/sctp_data_channel.cc b/pc/sctp_data_channel.cc
index f59dd1e..c63f820 100644
--- a/pc/sctp_data_channel.cc
+++ b/pc/sctp_data_channel.cc
@@ -309,12 +309,8 @@
   // so just add to the end of the queue and keep waiting.
   if (!queued_send_data_.Empty()) {
     if (!QueueSendDataMessage(buffer)) {
-      RTC_LOG(LS_ERROR) << "Closing the DataChannel due to a failure to queue "
-                           "additional data.";
-      // https://w3c.github.io/webrtc-pc/#dom-rtcdatachannel-send step 5
-      // Note that the spec doesn't explicitly say to close in this situation.
-      CloseAbruptlyWithError(RTCError(RTCErrorType::RESOURCE_EXHAUSTED,
-                                      "Unable to queue data for sending"));
+      // Queue is full
+      return false;
     }
     return true;
   }