Voice Communications

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Normally, when communicating with your neighbours during Line Operations, you are limited to text communication only. It is more realistic to simply press a button and call someone the way you would with a telephone. With the Nimajin Voice Communication Plugin, you can do just that, allowing you to speak directly with your neighbours without having to press buttons to convey your instructions. This frees up your hands for more dispatching work, and helps you to route traffic faster and more efficiently.


The following items are required:

  • Requires an up-to-date sim that can participate in Line Operations.
  • Broadband Internet connection of minimum 64Kbps (upload and download)
  • Functioning speakers and microphone.

It is highly recommended that if you do not already have one, that you obtain a headset with a microphone. Without one (and instead using speakers), the risk of annoying feedback loops are much higher.


The installation of the Voice Communication Plugin is designed to be simple. Even if you already have multiple sims installed, you only need to install the plugin once. It's recommended, though not required, that you install to the default installation directory, which is already chosen for you when you run the installer.

Ready to install...
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The default install location is already entered for you.
Installation complete!


Each sim maintains separate program settings for the plugin. The configuration steps below apply to every sim you already have installed (if you want to be able to use the plugin properly).

Before joining other dispatchers over the Internet, some settings need to be adjusted. To view and change these settings, access the Settings window of one of the simulations. If the plugin is installed correctly, a new settings page will appear at the bottom, below the "Language" entry: Select Voice to view the Voice Communications Plugin settings.


Choose which microphone you wish to use when talking to other dispatchers - it's possible to have more than one installed.


Choose which speakers (or headphones) you wish to use when listening to other dispatchers.


The plugin supports two levels audio quality, Normal, and High. The High level quality can reproduce sounds more clearly, but uses more bandwidth than the Normal level.

Normal typically requires 64Kbps up/down, while High requires twice that: 128Kbps up/down.


The Voice Communications Plugin requires 3 (three) non-overlapping, open UDP ports for each simulation. They are in a sequential block from N to N+2. In the example below, the simulation uses ports 55397, 55398 and 55399. Only the first port number needs to be entered - the next two port numbers are calculated automatically.

You will probably need to modify your port-forwarding settings on your router - see Multiplayer Ports and Forwarding for more detailed instructions.

Warning If you have, for example, 4 simulations and you wish to use the plugin with each of them, you must configure a total of 4 x 3 = 12 ports, three for each of the four sims. These ports must not interfere with other ports you may be using for Line Operations.

Suggestion The Simulation Network In Detail page has some excellent plans for configuring large numbers of sims for network play. If you plan to configure more than one or two sims for use with Line Operations and Voice Communication, this is the page for you.

Once you have set up port forwarding, press each of the Test Port buttons to see if you have forwarded the ports properly.

Voice Configuration Settings Page


When the Voice Communication Plugin is installed, there are few visible differences when running a simulation.

Status Bar Icon

One major difference is the addition of a "phone" icon in the status bar.

Icon Meaning Description
inline Default The default state - no one is calling you nor are you calling anyone else.
inline Connecting (or Disconnecting) A call is in the process of being connected - your partner has not yet answered your call, or someone is calling you (but you have not yet answered). May also appear briefly when ending a call.
inline Connected Call has been successfully connected - you should be able to hear your partner, and he/she should be able to hear you.

Placing a Call

Calling Neighbours

When you're ready to call a neighbour, simply press the button for the neighbour you wish to call, followed by the Call button. The process is exactly the same as calling an in-sim AI neighbour. To review, visit the page on the communications panel.

Calling Dispatchers via Telephone

In addition to calling immediate neighbours, you may also call dispatchers using the Telefon button. Both you and the person you're calling need to have the plugin installed and working in order to connect a voice call. If the person you are calling does not have the plugin installed correctly, communication will revert to the traditional buttons and textboxes, so you can still communicate, though not with voice. To review, visit the telephone page.


As soon as you've pressed the "Ruf" (Call) button, the inline icon should change to inline. When the callee (person being called) answers, the icon will change to inline and you should here his or her voice, and you may of course begin talking.

Answering a Call

When someone else is calling you your inline icon will change to inline. When you press the communication button for the calling neighbour, you should immediately say "Here Dispatcher ______" to confirm to your neighbour they have reached the correct person. Also, the inline should appear, indicating that the call is now connected.

Ending a Call

Simply press the "Ende" communication button. Either person may end a call at any time (exactly the same as with a telephone).


The publisher of the plugin may issue updates. When this happens, a notification will appear on the main Start Screen explaining what is available, together with a button Download Update to begin the update process.

Voip update 01.JPG

The new, updated plugin is downloaded and installed automatically. While you wait, a dialog box will appear to let you know what's happening. The entire program will then restart once the updating process was successful.

Voip update 02.JPG

Warning Depending on your Internet connection and the nature of the update, the process might take a while. Please be patient.

Technical Information


The plugin uses Voice over Internet Protocol, specifically, Session Initiation Protocol (SIP) as defined by RFC 3261.


All network traffic is direct, no intermediate servers are used, that is, all messages and media go directly from your router to the other player's router. Likewise, when playing on a LAN, all traffic goes directly from your machine to the other one, without going to the Internet.

Network Bandwidth Usage

You may need to consider your available bandwidth when choosing an audio transfer mode.

Network bandwidth is the capacity of your equipment and wiring to carry data. It is measured in bits per second. LANs provide bandwidth of 10 or 100 Mbps, WiFi is 3Mbps. Some equipment have different capacities in different directions. For example, a DSL connection might provide about 5 Mbps download bandwidth and 500 Kbps up. That means a person can send less data than he/she can receive in the same amount of time.

For Voice Communications, network traffic goes both ways. There is a continuous stream of data being sent from your computer to the other player, and the other player is sending a continuous stream of audio data to you. Both computers send the same amount of audio data at the same rate. There is no data sent while not talking.

The Voice Communications software uses a codec algorithm to compress the audio data before sending it so that it uses less network bandwidth. The codecs used are G.711, and in Normal mode, this is the same as used by the landline telephone network (ISDN). G.711 compresses the data to half of its original size. G.711 delivers by far the best audio quality of any 8 or 16 KHz sampling rate codec. You do not lose dynamic range or frequency fidelity. What you lose is half of the intermediate steps between silence and full loudness.

Normal quality audio mode samples audio 8,000 times per second to carry frequencies up to 4,000 Hz. High quality mode samples 16,000 times per second to carry frequencies up to 8,000 Hz. Skype, for example, samples at the same rate as High quality mode but compresses the audio with less fidelity.

A Normal mode stream requires bandwidth of 64 Kbps. A High quality mode stream requires 128 Kbps.

If you are connecting to another player on your LAN, you will have no problems using High quality mode.

If you are connected to the Internet by dialup, Voice Communications will not work. Phone modems cannot transfer enough audio data.

It's only when you connect to someone outside on the Internet that you may want to reduce to Normal mode. If your equipment or service won't support 128 Kbps upload, your partner will notice distorted or missing audio.

With asymmetrical DSL, the limiting factor is the lower upload rate. While set to High quality mode, the 128 Kbps required could be using half of your upload bandwidth.

You can check your Internet service capacity by finding a web site that offers a speed test, such as SpeedTest or BandwidthPlace.

Bandwidth will also depend on the distance of your connection. There may be slow equipment somewhere between you and your partner.

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