Rx_queue value in /proc/net/udp grows to twice the SO_RCVBUF value

TCP is backed by acks and retries to make sure you data gets where it's going. UDP is connectionless and "fire and forget". UDP is mostly used for streaming type applications, where if you lose some data you don't need to try to send it again.

Which one you use depends on the application. For example, a web server uses TCP.

You can safely reduce the time down, but you may run into issues with inproperly closed connections on networks with packet loss or jitter. I wouldn't start tuning at 1 second, start at 15-30 and work your way down.

Also, you really need to fix your application.

RFC 1185 has a good explanation in section 3.2:

When a TCP connection is closed, a delay of 2*MSL in TIME-WAIT state ties up the socket pair for 4 minutes (see Section 3.5 of [Postel81]. Applications built upon TCP that close one connection and open a new one (e.g., an FTP data transfer connection using Stream mode) must choose a new socket pair each time. This delay serves two different purposes:

 (a)  Implement the full-duplex reliable close handshake of TCP. 

      The proper time to delay the final close step is not really 
      related to the MSL; it depends instead upon the RTO for the 
      FIN segments and therefore upon the RTT of the path.* 
      Although there is no formal upper-bound on RTT, common 
      network engineering practice makes an RTT greater than 1 
      minute very unlikely.  Thus, the 4 minute delay in TIME-WAIT 
      state works satisfactorily to provide a reliable full-duplex 
      TCP close.  Note again that this is independent of MSL 
      enforcement and network speed. 

      The TIME-WAIT state could cause an indirect performance 
      problem if an application needed to repeatedly close one 
      connection and open another at a very high frequency, since 
      the number of available TCP ports on a host is less than 
      2**16.  However, high network speeds are not the major 
      contributor to this problem; the RTT is the limiting factor 
      in how quickly connections can be opened and closed. 
      Therefore, this problem will no worse at high transfer 
      speeds. 

 (b)  Allow old duplicate segements to expire. 

      Suppose that a host keeps a cache of the last timestamp 
      received from each remote host.  This can be used to reject 
      old duplicate segments from earlier incarnations of the 

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*Note: It could be argued that the side that is sending a FIN knows what degree of reliability it needs, and therefore it should be able to determine the length of the TIME-WAIT delay for the FIN's recipient. This could be accomplished with an appropriate TCP option in FIN segments.

      connection, if the timestamp clock can be guaranteed to have 
      ticked at least once since the old conennection was open. 
      This requires that the TIME-WAIT delay plus the RTT together 
      must be at least one tick of the sender's timestamp clock. 

      Note that this is a variant on the mechanism proposed by 
      Garlick, Rom, and Postel (see the appendix), which required 
      each host to maintain connection records containing the 
      highest sequence numbers on every connection.  Using 
      timestamps instead, it is only necessary to keep one quantity 
      per remote host, regardless of the number of simultaneous 
      connections to that host.