Voice Communications: Difference between revisions

Revision as of 17:04, 19 December 2012

UNDER CONSTRUCTION

Not a whole lot I can say: I don't know how they're integrating

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 (add link)

Ports

Extended range

(put link to multiplayer)

Forwarding

Traffic

All traffic is direct, no servers are used

Over the Internet all messages and media go direct from your router to the other router

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

Updates

To user, updates are automatic but should be prompted

Before loading VCP component, Sim should call updater, updater can look for update and return value if new available

Then sim should prompt user to install update

If user accepts then sim call updater to download and replace

VoIP

What is VoIP?

Yes, it's standard RFC3261 SIP

No registration is needed
Address lookup is provided by signalsoft multiplayer server

Bandwidth used

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

Usage

Communications panel?
Calling a Neighbour

Telephone panel?

Logic

Ie. How game is affected
Change in buttons

@@ Line 31: / Line 31: @@
 <p>No traffic while not talking?</p>
 <p>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.</p>
-<p>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.<p/>
+<p>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.</p>
 <p>LANs are 10 or 100 Mbps<br />
     WiFi is 3Mbps<br />