diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index d36c5985e5..e090fcf62f 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -1935,7 +1935,7 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
       codec_pair_id_, engine()->audio_jitter_buffer_max_packets_,
       engine()->audio_jitter_buffer_fast_accelerate_,
       engine()->audio_jitter_buffer_min_delay_ms_, unsignaled_frame_decryptor_,
-      crypto_options_, nullptr);
+      crypto_options_, unsignaled_frame_transformer_);
 
   recv_streams_.insert(std::make_pair(
       ssrc, new WebRtcAudioReceiveStream(std::move(config), call_)));
@@ -2481,6 +2481,13 @@ void WebRtcVoiceMediaChannel::SetDepacketizerToDecoderFrameTransformer(
     uint32_t ssrc,
     rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
   RTC_DCHECK_RUN_ON(worker_thread_);
+  if (ssrc == 0) {
+    // If the receiver is unsignaled, save the frame transformer and set it when
+    // the stream is associated with an ssrc.
+    unsignaled_frame_transformer_ = std::move(frame_transformer);
+    return;
+  }
+
   auto matching_stream = recv_streams_.find(ssrc);
   if (matching_stream == recv_streams_.end()) {
     RTC_LOG(LS_INFO) << "Attempting to set frame transformer for SSRC:" << ssrc
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index 9cb7ec82eb..0a501bea0a 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -321,6 +321,8 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   // Unsignaled streams have an option to have a frame decryptor set on them.
   rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
       unsignaled_frame_decryptor_;
+  rtc::scoped_refptr<webrtc::FrameTransformerInterface>
+      unsignaled_frame_transformer_;
 };
 }  // namespace cricket