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Basics: Network Bottlenecks

At the network level, many things can affect performance. The bandwidth (the amount of data that can be carried by the network) tends to be the first culprit checked. Assuming you have determined that bad performance is attributable to the network component of an application, there is more likely cause of bad network performance than network bandwidth. The most likely cause of bad network performance is the application itself and how it is handling distributed data and functionality.

The overall speed of a particular network connection is limited by the slowest link in the connection chain and the length of the chain. Identifying the slowest link is difficult and may not even be consistent: it can vary at different times of the day or for different communication paths. A network communication path lead from an application through a TCP/IP stack (which adds various layers of headers, possibly encrypting and compressing data as well), then through the hardware interface, through a modem, over a phone line, through another modem, over to a service provider’s router, through many heavily congested data lines of various carrying capacities and multiple routers with different maximum throughputs and configurations, to a machine at the other end with its own hard interface, TCP/IP stack and application. A typical web download route is just like this. In addition, there are dropped packets, acknowledgements, retries, bus contention, and so on.

Because so many possibilities causes of bad network performance are external to an application, one option you can consider including in an application is a network speed testing facility that reports to the user. This should test the speed of data transfer from the machine to various destinations: to itself, to another machine on the local network, to the Internet Service Provider, to the target server across the network, and to any other destinations appropriate. This type of diagnostics report can tell users that they are obtaining bad performance from something other than your application. If you feel that the performance of your application is limited by the actual network communication speed, and not by other (application) factors, this facility will report the maximum possible speeds to your user.


Latency is different from the load-carrying capacity (bandwidth) of a network. Bandwidth refers to how much data can be sent down the communication channel for a given period of time and is limited by the link in the communication chain that has the lowest bandwidth. The latency is the amount of time a particular data packet takes to get from one end of the communication channel to the other. Bandwidth tells you the limits within which your application can operate before the performance become affected by the volume of data being transmitted. Latency often affects the user’s view of the performance even when bandwidth isn’t a problem.

In most cases, especially Internet traffic, latency is an important concern. You can determine the basic round-trip time for a data packets from any two machines using the ping utility. This utility provides a measure of the time it takes a packet of data to reach another machine and be returned. However, the time measure is for a basic underlying protocol (ICMP packet) to travel between the machines. If the communication channel is congested and the overlying protocol requires re-transmissions (often the case for Internet traffic), one transmission at the application level can actually be equivalent to many round trips.

It is important to be aware of these limitations. It is often possible to tune the application to minimize the number of transfers by packing data together, caching and redesigning the distributed application protocol to aim for a less conversational mode of operation. At the network level, you need to monitor the transmission statistics (using the ping and netstat utilities and packet sniffers) and consider tuning any network parameters that you have access to in order to reduce re-transmissions.

TCP/IP Stacks

The TCP/IP stack is the section of code that is responsible for translating each application-level network request (send, receive, connect, etc.) through the transport layers down to the wire and back up to the application at the other end of the connection. Because the stacks are usually delivered with the operation system and performance-tested before delivery (since a slow network connection on an otherwise fast machine and fast network is pretty obvious), it is unlikely that the TCP/IP stack itself is a performance problem.

In addition to the stack itself, stacks include several tunable parameters. Most of these parameters deal with transmission details beyond the scope of the book. One parameter worth mentioning is the maximum packet size. When your application sends data, the underlying protocol breaks the data into packets that are transmitted. There is an optimal size for packets transmitted over a particular communication channel, and the packet size actually used by the stack is compromise. Smaller packets are less likely to be dropped, but they introduced more overhead, as data probably has to be broken up into more packets with more header overhead.

If your communication takes place over a particular set of endpoints, you may want to alter the packet sizes. For a LAN segment with no router involved, the packets can be big (e.g. 8KB). For a LAN with routers, you probably want to set the maximum packet size to the size the routers allow to pass unbroken. (Routers can break up the packets into smaller ones; 1500 bytes is the typical maximum packet size and the standard for the Ethernet. The maximum packet size is configurable by the router’s network administrator.) If your application is likely to be sending data over the Internet and you cannot guarantee the route and quality of routers it will pass through, 500 bytes per packet is likely to be optimal.

Network Bottlenecks

Other causes of slow network I/O can be attributed directly to the load or configuration of the network. For example, a LAN may become congested when many machines are simultaneously trying to communicate over the network. The potential throughput of the network could handle the load, but the algorithms to provide communication channels slow the network, resulting in a lower maximum throughput. A congested Ethernet network has an average throughput approximately one third the potential maximum throughputs. Congested networks have other problems, such as dropped network packets. If you are using TCP, the communication rate on a congested network is much slower as the protocol automatically resends the dropped packets. If you are using UDP, your application must resend multiple copies for each transfer. Dropping packets in this way is common for the Internet. For LANs, you need to coordinate closely with network administrators to alert them to the problem. For single machines connected by a service provider, suggesting improvements. The phone line to the service provider may be noisier than expected: if so, you also need to speak to the phone line provider. It is also worth checking with the service provider, who should have optimal configurations they can demonstrate.

Dropped packets and re-transmissions are a good indication of network congestion problems, and you should be on constant lookup for them. Dropped packets often occur when routers are overloaded and find it necessary to drop some of the packets being transmitted as the router’s buffer overflow. This means that the overlying protocol will request the packets to be resent. The netstat utility lists re-transmission and other statistics that can identify these sorts of problems. Re-transmissions may indicate that the maximum packet size is too large.

DNS Lookup

Looking up network address is an often-overlooked cause of bad network performance. When your application tries to connect to a network address such as (e.g. downloading a webpage from, your application first translates into a four-byte network IP address such as This is the actual address that the network understands and uses for routing network packets. The is this translation works is that your system is configured with some seldom-used files that can specify this translation, and a more frequently used Domain Name System (DNS) server that can dynamically provide you with the address from the given string. DBS translation works as follows:

  1. The machine running the application sends the text string of the hostname (e.g. to the DNS server.
  2. The DNS server checks its cache to find an IP address corresponding to that hostname. If the server does not find an entry in the cache, it asks its own DNS server (usually further up the Internet domain-name hierarchy) until ultimately the name is resolved. (This may be by components of the name being resolved, e.g. first .org, then, etc, each time asking another machine as the search request is successively resolved.) This resolved IP address is added to the DBS server’s cache.
  3. The IP address s returned to the original machine running the application.
  4. The application uses the IP address to connect to the desired destination.

The address lookup does not need to be repeated once a connection is established, but any other connections (within the same session of the application or in other session s at the same time and later) need to repeat the lookup procedure to start another connection.

You can improve this situation by running a DNS server locally on the machine, or on a local server if the application uses a LAN. A DNS server can be run as a “caching-only” server that resets its cache each time the machine is rebooted. There would be little point in doing this if the machine used only one or two connections per hostname between successive reboots. For more frequent connections, a local DNS server can provide a noticeable speedup to connections. Nslookup is useful for investigating how a particular system does translations.

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