diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
index 26152df0f0..2bb71941f9 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
@@ -23,6 +23,12 @@
 namespace webrtc {
 namespace {
 
+// Using a reasonable default of 10ms for the retransmission delay for frames
+// not coming from this sender's encoder. This is usually taken from an
+// estimate of the RTT of the link,so 10ms should be a reasonable estimate for
+// frames being re-transmitted to a peer, probably on the same network.
+const TimeDelta kDefaultRetransmissionsTime = TimeDelta::Millis(10);
+
 class TransformableVideoSenderFrame : public TransformableVideoFrameInterface {
  public:
   TransformableVideoSenderFrame(const EncodedImage& encoded_image,
@@ -209,15 +215,17 @@ void RTPSenderVideoFrameTransformerDelegate::SendVideo(
     auto* transformed_video_frame =
         static_cast<TransformableVideoFrameInterface*>(transformed_frame.get());
     VideoFrameMetadata metadata = transformed_video_frame->Metadata();
-    sender_->SendVideo(
-        transformed_video_frame->GetPayloadType(), metadata.GetCodec(),
-        transformed_video_frame->GetTimestamp(),
-        /*capture_time=*/Timestamp::MinusInfinity(),
-        transformed_video_frame->GetData(),
-        transformed_video_frame->GetData().size(),
-        RTPVideoHeader::FromMetadata(metadata),
-        /*expected_retransmission_time=*/TimeDelta::PlusInfinity(),
-        metadata.GetCsrcs());
+    // TODO(bugs.webrtc.org/14708): Use an actual RTT estimate for the
+    // retransmission time instead of a const default, in the same way as a
+    // locally encoded frame.
+    sender_->SendVideo(transformed_video_frame->GetPayloadType(),
+                       metadata.GetCodec(),
+                       transformed_video_frame->GetTimestamp(),
+                       /*capture_time=*/Timestamp::MinusInfinity(),
+                       transformed_video_frame->GetData(),
+                       transformed_video_frame->GetData().size(),
+                       RTPVideoHeader::FromMetadata(metadata),
+                       kDefaultRetransmissionsTime, metadata.GetCsrcs());
   }
 }
 
@@ -254,13 +262,14 @@ std::unique_ptr<TransformableVideoFrameInterface> CloneSenderVideoFrame(
                                  ? VideoFrameType::kVideoFrameKey
                                  : VideoFrameType::kVideoFrameDelta;
   // TODO(bugs.webrtc.org/14708): Fill in other EncodedImage parameters
-
+  // TODO(bugs.webrtc.org/14708): Use an actual RTT estimate for the
+  // retransmission time instead of a const default, in the same way as a
+  // locally encoded frame.
   VideoFrameMetadata metadata = original->Metadata();
   RTPVideoHeader new_header = RTPVideoHeader::FromMetadata(metadata);
   return std::make_unique<TransformableVideoSenderFrame>(
       encoded_image, new_header, original->GetPayloadType(), new_header.codec,
-      original->GetTimestamp(),
-      /*expected_retransmission_time=*/TimeDelta::PlusInfinity(),
+      original->GetTimestamp(), kDefaultRetransmissionsTime,
       original->GetSsrc(), metadata.GetCsrcs());
 }
 
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc
index ad00e6449f..6790fc3a71 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate_unittest.cc
@@ -83,7 +83,7 @@ class RtpSenderVideoFrameTransformerDelegateTest : public ::testing::Test {
     delegate->TransformFrame(
         /*payload_type=*/1, VideoCodecType::kVideoCodecVP8, /*rtp_timestamp=*/2,
         encoded_image, RTPVideoHeader::FromMetadata(metadata),
-        /*expected_retransmission_time=*/TimeDelta::PlusInfinity());
+        /*expected_retransmission_time=*/TimeDelta::Millis(10));
     return frame;
   }
 
@@ -123,7 +123,7 @@ TEST_F(RtpSenderVideoFrameTransformerDelegateTest,
   delegate->TransformFrame(
       /*payload_type=*/1, VideoCodecType::kVideoCodecVP8, /*rtp_timestamp=*/2,
       encoded_image, RTPVideoHeader(),
-      /*expected_retransmission_time=*/TimeDelta::PlusInfinity());
+      /*expected_retransmission_time=*/TimeDelta::Millis(10));
 }
 
 TEST_F(RtpSenderVideoFrameTransformerDelegateTest,
@@ -260,7 +260,7 @@ TEST_F(RtpSenderVideoFrameTransformerDelegateTest,
       test_sender_,
       SendVideo(payload_type, absl::make_optional(kVideoCodecVP8), timestamp,
                 /*capture_time=*/Timestamp::MinusInfinity(), buffer, _, _,
-                /*expected_retransmission_time=*/TimeDelta::PlusInfinity(),
+                /*expected_retransmission_time=*/TimeDelta::Millis(10),
                 frame_csrcs))
       .WillOnce(WithoutArgs([&] {
         event.Set();
@@ -310,7 +310,7 @@ TEST_F(RtpSenderVideoFrameTransformerDelegateTest,
   delegate->TransformFrame(
       /*payload_type=*/1, VideoCodecType::kVideoCodecVP8, /*rtp_timestamp=*/2,
       encoded_image, RTPVideoHeader(),
-      /*expected_retransmission_time=*/TimeDelta::PlusInfinity());
+      /*expected_retransmission_time=*/TimeDelta::Millis(10));
 }
 
 }  // namespace