fix ErrShortBuffer error by increasing the temporary buffer size

Refs webrtc-rs/sctp#28
melekes · Jul 11, 2022 · 64d60e9 · 64d60e9
1 parent 6d703f8
commit 64d60e9
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Showing 2 changed files with 10 additions and 2 deletions.
diff --git a/transports/webrtc/src/transport.rs b/transports/webrtc/src/transport.rs
@@ -502,7 +502,11 @@ async fn do_dial(
     //     .await
     //     .map_err(|e| Error::WebRTC(e.into()))?;
 
-    Ok((peer_id, Connection::new(peer_connection).await))
+    let mut c = Connection::new(peer_connection).await;
+    // XXX: default buffer size is too small to fit some messages. Possibly remove once
+    // https://github.com/webrtc-rs/sctp/issues/28 is fixed.
+    c.set_data_channels_read_buf_capacity(8192 * 10);
+    Ok((peer_id, c))
 }
 
 /// Creates a [`Multiaddr`] from the given IP address and port number.

diff --git a/transports/webrtc/src/upgrade.rs b/transports/webrtc/src/upgrade.rs
@@ -160,5 +160,9 @@ pub async fn webrtc(
     //     .await
     //     .map_err(|e| Error::WebRTC(e.into()))?;
 
-    Ok((peer_id, Connection::new(peer_connection).await))
+    let mut c = Connection::new(peer_connection).await;
+    // XXX: default buffer size is too small to fit some messages. Possibly remove once
+    // https://github.com/webrtc-rs/sctp/issues/28 is fixed.
+    c.set_data_channels_read_buf_capacity(8192 * 10);
+    Ok((peer_id, c))
 }