Voice Communications

UNDER CONSTRUCTION

Not a whole lot I can say: I don't know how Signalsoft is integrating this in to the simulations' UI.

Previously in the simulations with Dispatcher Centre installed, you could communicate with other players on the network, but only by sending text messages. I guarantee you that not many real dispatchers texted each other in the 1900's! Now, with a headset and the Voice Communications Plugin, you can hear and talk to your fellow dispatchers. And your gameplay is not distracted by all that typing. The sound is clear and the networking does not conflict with any VoIP you may have installed.

Integration?

Requires multiplayer Dispatcher Supervisor. Describe how Voice Communications appears in the simulation UI.

Usage

How game is affected?
Ie. Change in buttons

Calling a Neighbour:

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

Port Forwarding

Voice Communications requires 3 more 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

Traffic

All network traffic is direct, no intermediate servers are used.

Over the Internet all messages and media go directly from your router to the other player's router.

On LAN, all traffic goes directly from your machine to the other one, without going to the Internet.

VoIP

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.

Network Bandwidth Usage

Traffic goes both ways

No traffic while not talking?

The 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 uLaw, the same as used by the landline telephone network. 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 samples 8,000 times per second to carry frequencies up to 4,000 Hz. Enhanced (name?) samples 16,000 times per second to carry frequencies up to 8,000 Hz. Skype samples at enhanced rate but uses a more lossy codec. A Normal stream requires bandwidth of 64 Kbps. An Enhanced stream requires 128 Kbps.

LANs are 10 or 100 Mbps
WiFi is 3Mbps
My DSL internet is 4.6 to 5.3 Mbps down and 310 to 740 Kbps up
So limiter is upload:
I could be using 1/2 my upload bandwidth while talking 128 Kbps

LAN traffic is direct, not through router

Updates

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.