diff --git a/api/media_transport_interface.cc b/api/media_transport_interface.cc
index de04e197fa..abd9861652 100644
--- a/api/media_transport_interface.cc
+++ b/api/media_transport_interface.cc
@@ -88,4 +88,11 @@ size_t MediaTransportInterface::GetAudioPacketOverhead() const {
 void MediaTransportInterface::SetAllocatedBitrateLimits(
     const MediaTransportAllocatedBitrateLimits& limits) {}
 
+// TODO(mellem):  Delete when all implementations support it.
+RTCError MediaTransportInterface::OpenChannel(int channel_id) {
+  // NB: This must return OK to avoid breaking existing implementations, which
+  // do not require calling OpenChannel.
+  return RTCError::OK();
+}
+
 }  // namespace webrtc
diff --git a/api/media_transport_interface.h b/api/media_transport_interface.h
index 2f5431f980..edbd3b1da0 100644
--- a/api/media_transport_interface.h
+++ b/api/media_transport_interface.h
@@ -278,6 +278,11 @@ class MediaTransportInterface {
   virtual void SetAllocatedBitrateLimits(
       const MediaTransportAllocatedBitrateLimits& limits);
 
+  // Opens a data |channel_id| for sending.  May return an error if the
+  // specified |channel_id| is unusable.  Must be called before |SendData|.
+  // TODO(mellem):  Make pure virtual when all implementations support it.
+  virtual RTCError OpenChannel(int channel_id);
+
   // Sends a data buffer to the remote endpoint using the given send parameters.
   // |buffer| may not be larger than 256 KiB. Returns an error if the send
   // fails.
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 341a570b68..0288afca3b 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -5694,7 +5694,7 @@ void PeerConnection::DisconnectDataChannel(DataChannel* webrtc_data_channel) {
 
 void PeerConnection::AddSctpDataStream(int sid) {
   if (media_transport_) {
-    // No-op.  Media transport does not need to add streams.
+    media_transport_->OpenChannel(sid);
     return;
   }
   if (!sctp_transport_) {