diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 81c517ab31..be1f3a778a 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -1877,7 +1877,7 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
   if (role() == MediaChannel::Role::kBoth) {
     // In legacy kBoth mode, the MediaChannel takes the responsibility for
     // telling the receiver about the local SSRC.
-    // In kSend mode, this is the caller's responsibility.
+    // In kSend mode, this happens via a callback.
 
     // At this point the stream's local SSRC has been updated. If it is the
     // first send stream, make sure that all the receive streams are updated
@@ -1888,6 +1888,12 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
         call_->OnLocalSsrcUpdated(kv.second->stream(), ssrc);
       }
     }
+  } else if (ssrc_list_changed_callback_) {
+    std::set<uint32_t> ssrcs_in_use;
+    for (auto it : send_streams_) {
+      ssrcs_in_use.insert(it.first);
+    }
+    ssrc_list_changed_callback_(ssrcs_in_use);
   }
 
   send_streams_[ssrc]->SetSend(send_);
diff --git a/pc/peer_connection_mediachannel_split_unittest.cc b/pc/peer_connection_mediachannel_split_unittest.cc
index a6b9640269..97e5ecdc40 100644
--- a/pc/peer_connection_mediachannel_split_unittest.cc
+++ b/pc/peer_connection_mediachannel_split_unittest.cc
@@ -173,6 +173,32 @@ TEST_P(PeerConnectionMediaChannelSplitTest, VideoPacketLossCausesNack) {
   EXPECT_TRUE_WAIT(NacksReceivedCount(*caller()) > 0, kDefaultTimeout);
 }
 
+// Test that we can get capture start ntp time.
+TEST_P(PeerConnectionMediaChannelSplitTest,
+       GetCaptureStartNtpTimeWithOldStatsApi) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->AddAudioTrack();
+
+  callee()->AddAudioTrack();
+
+  // Do offer/answer, wait for the callee to receive some frames.
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+
+  // Get the remote audio track created on the receiver, so they can be used as
+  // GetStats filters.
+  auto receivers = callee()->pc()->GetReceivers();
+  ASSERT_EQ(1u, receivers.size());
+  auto remote_audio_track = receivers[0]->track();
+
+  // Get the audio output level stats. Note that the level is not available
+  // until an RTCP packet has been received.
+  EXPECT_TRUE_WAIT(callee()->OldGetStatsForTrack(remote_audio_track.get())
+                           ->CaptureStartNtpTime() > 0,
+                   2 * kMaxWaitForFramesMs);
+}
+
 INSTANTIATE_TEST_SUITE_P(PeerConnectionMediaChannelSplitTest,
                          PeerConnectionMediaChannelSplitTest,
                          Values("WebRTC-SplitMediaChannel/Disabled/",