Do not prune if the current best connection is weak.

webrtc/p2p/base/port.h

 /*
  *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 418 matching lines @@
   };
 
   WriteState write_state() const { return write_state_; }
   bool writable() const { return write_state_ == STATE_WRITABLE; }
   bool receiving() const { return receiving_; }
 
   // Determines whether the connection has finished connecting.  This can only
   // be false for TCP connections.
   bool connected() const { return connected_; }
 
+  bool Weak() const { return !(writable() && receiving() && connected()); }
+
   // Estimate of the round-trip time over this connection.
   uint32 rtt() const { return rtt_; }
 
   size_t sent_total_bytes();
   size_t sent_bytes_second();
   // Used to track how many packets are discarded in the application socket due
   // to errors.
   size_t sent_discarded_packets();
   size_t sent_total_packets();
   size_t recv_total_bytes();
@@ skipping 178 matching lines @@
                    const rtc::PacketOptions& options);
   virtual int GetError() { return error_; }
 
  private:
   int error_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_P2P_BASE_PORT_H_