Voice Communications: Difference between revisions
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<p>You may need to consider your available bandwidth when choosing an audio transfer mode.</p> | <p>You may need to consider your available bandwidth when choosing an audio transfer mode.</p> | ||
<p>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.</p> | <p>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.</p> | ||
<p>For Voice Communications 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 audio data at the same rate. | <p>For Voice Communications 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.</p> | ||
<p>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.</p> | |||
<p>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, 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 mode samples audio 8,000 times per second to carry frequencies up to 4,000 Hz. Enhanced (correct name?) mode samples 16,000 times per second to carry frequencies up to 8,000 Hz. Skype samples at Enhanced rate but compresses the audio with less fidelity. A Normal stream requires bandwidth of 64 Kbps. An Enhanced stream requires 128 Kbps.</p> | <p>Normal mode samples audio 8,000 times per second to carry frequencies up to 4,000 Hz. Enhanced (correct name?) mode samples 16,000 times per second to carry frequencies up to 8,000 Hz. Skype samples at Enhanced rate but compresses the audio with less fidelity. A Normal stream requires bandwidth of 64 Kbps. An Enhanced stream requires 128 Kbps.</p> | ||
<p> | <p>If you are connecting to another player on your LAN, you will have no problems using Enhanced mode.</p> | ||
<p>If you're connected to the Internet by dialup, Voice Communications will not work. Phone modems cannot transfer enough audio data.</p> | |||
<p>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.</p> | |||
<p>In my case, with asymmetrical DSL, the limiting factor is the lower upload rate. While set to Enhanced mode, the 128 Kbps required could be using half of my upload bandwidth. Normal mode works well.</p> | |||
<p>You can check your Internet service capacity by finding a web site that offers a speed test, such as [http://www.speedtest.net/ SpeedTest] or [http://www.bandwidthplace.com BandwidthPlace].</p> | |||
<p>Bandwidth will also depend on the distance of your connection. There may be slow equipment somewhere between you and your partner.</p> | |||
<h3>Updates</h3> | <h3>Updates</h3> | ||
<p>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.</p> | <p>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.</p> | ||
{{EnglishNavBottom}} | {{EnglishNavBottom}} | ||
Revision as of 19:24, 19 December 2012
Contents | FAQ | Manual | General Tips and Tricks | Change Log | Multiplayer Manual | Line Operations Manual | Voice Communications | Change Log | Developers pages
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.
Usage
Requires multiplayer Dispatcher Supervisor. Describe how Voice Communications appears in the simulation UI.
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
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.
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.
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 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 mode samples audio 8,000 times per second to carry frequencies up to 4,000 Hz. Enhanced (correct name?) mode samples 16,000 times per second to carry frequencies up to 8,000 Hz. Skype samples at Enhanced rate but compresses the audio with less fidelity. A Normal stream requires bandwidth of 64 Kbps. An Enhanced stream requires 128 Kbps.
If you are connecting to another player on your LAN, you will have no problems using Enhanced 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 Enhanced 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.
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.