webrtc: increase receive buffer size on listener (#2730)

p2p/transport/webrtc/listener.go

@@ -204,6 +204,10 @@ func (l *listener) setupConnection(
  l.transport.peerConnectionTimeouts.Failed,
  l.transport.peerConnectionTimeouts.Keepalive,
  )
+ // This is higher than the path MTU due to a bug in the sctp chunking logic.
+ // Remove this after https://github.com/pion/sctp/pull/301 is included
+ // in a release.
+ settingEngine.SetReceiveMTU(udpmux.ReceiveBufSize)
  settingEngine.DetachDataChannels()
 
  w, err = newWebRTCConnection(settingEngine, l.config)
@@ -314,14 +318,13 @@ func (l *listener) Multiaddr() ma.Multiaddr {
 }
 
 // addOnConnectionStateChangeCallback adds the OnConnectionStateChange to the PeerConnection.
-// The channel returned here:
-// * is closed when the state changes to Connection
-// * receives an error when the state changes to Failed
-// * doesn't receive anything (nor is closed) when the state changes to Disconnected
+// If the connection establishment errors, an error is written to the channel before closing.
+// If the connection establishment is successful, the channel is closed without writing anything.
 func addOnConnectionStateChangeCallback(pc *webrtc.PeerConnection) <-chan error {
  errC := make(chan error, 1)
  var once sync.Once
  pc.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
+ fmt.Println("connection state: ", state)
  switch state {
  case webrtc.PeerConnectionStateConnected:
  once.Do(func() { close(errC) })
@@ -336,6 +339,13 @@ func addOnConnectionStateChangeCallback(pc *webrtc.PeerConnection) <-chan error
  // If the connection then receives packets on the connection, it can move back to the connected state.
  // If no packets are received until the failed timeout is triggered, the connection moves to the failed state.
  log.Warn("peerconnection disconnected")
+ case webrtc.PeerConnectionStateClosed:
+ // ConnectionStateClosed is a terminal state. This happens when the peer closes the PeerConnection before
+ // connection establishment.
+ once.Do(func() {
+ errC <- errors.New("peerconnection closed")
+ close(errC)
+ })
  }
  })
  return errC

p2p/transport/webrtc/transport.go

@@ -78,7 +78,7 @@ const (
 const (
  DefaultDisconnectedTimeout = 20 * time.Second
  DefaultFailedTimeout = 30 * time.Second
- DefaultKeepaliveTimeout = 15 * time.Second
+ DefaultKeepaliveTimeout = 5 * time.Second
 )
 
 type WebRTCTransport struct {

p2p/transport/webrtc/transport_test.go

@@ -110,14 +110,16 @@ func TestTransportWebRTC_CanListenSingle(t *testing.T) {
 
  done := make(chan struct{})
  go func() {
- _, err := tr1.Dial(context.Background(), listener.Multiaddr(), listeningPeer)
+ conn, err := tr1.Dial(context.Background(), listener.Multiaddr(), listeningPeer)
  assert.NoError(t, err)
+ t.Cleanup(func() { conn.Close() })
  close(done)
  }()
 
  conn, err := listener.Accept()
  require.NoError(t, err)
  require.NotNil(t, conn)
+ defer conn.Close()
 
  require.Equal(t, connectingPeer, conn.RemotePeer())
  select {
@@ -170,12 +172,14 @@ func TestTransportWebRTC_CanListenMultiple(t *testing.T) {
  defer wg.Done()
  ctr, _ := getTransport(t)
  conn, err := ctr.Dial(ctx, listener.Multiaddr(), listeningPeer)
+ if conn != nil {
+ t.Cleanup(func() { conn.Close() })
+ }
  select {
  case <-ctx.Done():
  default:
  assert.NoError(t, err)
  assert.NotNil(t, conn)
- t.Cleanup(func() { conn.Close() })
  }
  }()
  }