Voice Communications

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Contents | FAQ | Manual | General Tips and Tricks | Change Log | Multiplayer Manual | Line Operations Manual | Voice Communications | Change Log | Developers pages



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 at bare minimum 64Kbps.
  • 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...
A brief message from our lawyers.
The default install location is already entered for you.
Installation complete!


Before joining other dispatchers over the Internet, some settings need to be adjusted. To view and change these settings,

Voice Configuration Settings Page

Port Forwarding

Voice Communications requires 3 UDP ports. They are in a sequential block from N to N+2. The simulations specify the beginning port number N. These should be within the range specified by multiplayer. You may have to extend the port range in your router and firewall.

See Multiplayer Ports and Forwarding


Describe changes to operation of Communications panel
Describe changes to operation of Telephone panel


Software updates are automatic after your Voice Communications Plugin is registered. When a simulation starts, it will check the Internet for new files and prompt you when new parts are available. Updates may be slow, because up to 8MB may be downloaded. The simulation will be restarted after the new files are installed.

Technical Information


What is VoIP? Yes, it's standard RFC3261 SIP With typical VoIP programs like Skype, you have to register with the system so other people can call you. No registration is needed here, address lookup is provided by Signalsoft multiplayer server. You're only visible to other dispatchers in the same Line Operation.


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, my DSL connection provides about 5 Mbps download bandwidth and 500 Kbps up. That means I can send less data than I 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 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. An 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're 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.

In my case, 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 my upload bandwidth. Normal mode works well. 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.

Contents | Sp Dr S 60 | FAQ | Manual | General Tips and Tricks | Multiplayer Manual | Line Operations Manual | Voice Communications | Change Log | Developers pages