SignalWiki - User contributions [en-gb]

Voice Communications

2013-01-07T20:47:06Z

Ted.Szoczei: /* Network Bandwidth Usage */

{{EnglishNav}}

<div style="background:#98FB98;padding:5px;border:1px solid #006400;">
UNDER CONSTRUCTION
</div>

=Purpose=

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.

=Prerequisites=

The following items are ''required'':
* Requires an up-to-date sim that can participate in [[Line Operations Manual|Line Operations]].
* Broadband Internet connection of ''at bare 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.

=Installation=

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.

{|
|[[File:Voip_install_01.JPG|left|450px|frameless|Ready to install...]]
|[[File:Voip_install_02.JPG|left|450px|frameless|A brief message from our lawyers.]]
|-
|[[File:Voip_install_03.JPG|left|450px|frameless|The default install location is already entered for you.]]
|[[File:Voip_install_04.JPG|left|450px|frameless|Installation complete!]]
|}

<div style="clear: both"></div> 

=Configuration=

<p style="display:inline-block;background:#fffa6a;padding:5px;border:1px solid #ffc600;">
'''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).
</p>

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.

==Microphone==

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

==Speakers==

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

==Quality==

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.

==Port==

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 5539'''7''', 5539'''8''' and 5539'''9'''. 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_Manual#Ports|Multiplayer Ports and Forwarding]] for more detailed instructions.

<p style="display:inline-block;background:#fffa6a;padding:5px;border:1px solid #ffc600;">
'''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.
</p>

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

[[File:Voip_config_02.JPG|left|Voice Configuration Settings Page]]

<div style="clear: both"></div> 

=Use=

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.

{| cellpadding="10" cellspacing="0" border="1"
|-
|'''Icon'''||'''Meaning'''||'''Description'''
|-
|[[File:Voip_icon_enabled.png|inline]]||Default||The default state - no one is calling you nor are you calling anyone else.
|-
|[[File:Voip_icon_transition.png|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.
|-
|[[File:Voip_icon_connected.png |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==

When you're reading 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 [[Communication|communications panel]].

As soon as you've pressed the "Ruf" (Call) button, the [[File:Voip_icon_enabled.png|inline]] icon should change to [[File:Voip_icon_transition.png|inline]]. When your neighbour answers, the icon will change to [[File:Voip_icon_connected.png |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 [[File:Voip_icon_enabled.png|inline]] icon will change to [[File:Voip_icon_transition.png|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 [[File:Voip_icon_connected.png |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).

=Updates=

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.

[[File: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.

[[File:Voip_update_02.JPG]]

<p style="display:inline-block;background:#fffa6a;padding:5px;border:1px solid #ffc600;">
'''Warning''' Depending on your Internet connection and the nature of the update, the process might take a while. Please be patient.
</p>

=Technical Information=
==VoIP==

The plugin uses [http://en.wikipedia.org/wiki/Voice_over_IP Voice over Internet Protocol], specifically, [http://en.wikipedia.org/wiki/Session_Initiation_Protocol Session Initiation Protocol (SIP)] as defined by RFC 3261.

==Traffic==

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 [http://www.speedtest.net/ SpeedTest] or [http://www.bandwidthplace.com BandwidthPlace].

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

{{EnglishNavBottom}}

Voice Communications

2012-12-20T03:40:06Z

Ted.Szoczei: Changed 'enhanced' to high quality'

{{EnglishNav}}

<div style="background:#98FB98;padding:5px;border:1px solid #006400;">
UNDER CONSTRUCTION
</div>
<p>Not a whole lot I can say: I don't know how Signalsoft is integrating this in to the simulations' UI.</p>
<p>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.</p>
<h3>Usage</h3>
<p>Requires multiplayer [[Line Operations Manual|Dispatcher Supervisor]]. Describe how Voice Communications appears in the simulation UI.</p>
<p>How game is affected?<br />
Ie. Change in buttons</p>
<h5>Calling a Neighbour</h5>
<p>Describe changes to operation of [[Communication|Communications panel]]<br />
Describe changes to operation of [[Line_Operations_Manual#Telephone|Telephone panel]]</p>

<h3>Port Forwarding</h3>
<p>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.</p>
<p>See [[Multiplayer_Manual#Ports|Multiplayer Ports and Forwarding]]</p>

<h3>VoIP</h3>
<p>What is VoIP?</p>
<p>Yes, it's standard RFC3261 SIP</p>
<p>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.</p>

<h3>Traffic</h3>
<p>All network traffic is direct, no intermediate servers are used.</p>
<p>Over the Internet all messages and media go directly from your router to the other player's router.</p>
<p>On LAN, all traffic goes directly from your machine to the other one, without going to the Internet.</p>

<h3>Network Bandwidth Usage</h3>
<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>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.</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>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.</p>
<p>A Normal mode stream requires bandwidth of 64 Kbps. An High quality mode stream requires 128 Kbps.</p>
<p>If you are connecting to another player on your LAN, you will have no problems using High quality 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 High quality 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>
<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}}

Voice Communications

2012-12-19T19:42:48Z

Ted.Szoczei:

{{EnglishNav}}

<div style="background:#98FB98;padding:5px;border:1px solid #006400;">
UNDER CONSTRUCTION
</div>
<p>Not a whole lot I can say: I don't know how Signalsoft is integrating this in to the simulations' UI.</p>
<p>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.</p>
<h3>Usage</h3>
<p>Requires multiplayer [[Line Operations Manual|Dispatcher Supervisor]]. Describe how Voice Communications appears in the simulation UI.</p>
<p>How game is affected?<br />
Ie. Change in buttons</p>
<h5>Calling a Neighbour</h5>
<p>Describe changes to operation of [[Communication|Communications panel]]<br />
Describe changes to operation of [[Line_Operations_Manual#Telephone|Telephone panel]]</p>

<h3>Port Forwarding</h3>
<p>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.</p>
<p>See [[Multiplayer_Manual#Ports|Multiplayer Ports and Forwarding]]</p>
<h3>VoIP</h3>

<p>What is VoIP?</p>
<p>Yes, it's standard RFC3261 SIP</p>
<p>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.</p>

<h3>Traffic</h3>
<p>All network traffic is direct, no intermediate servers are used.</p>
<p>Over the Internet all messages and media go directly from your router to the other player's router.</p>
<p>On LAN, all traffic goes directly from your machine to the other one, without going to the Internet.</p>

<h3>Network Bandwidth Usage</h3>
<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>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.</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>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.</p>
<p>A Normal mode stream requires bandwidth of 64 Kbps. An Enhanced mode stream requires 128 Kbps.</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>
<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}}

Voice Communications

2012-12-19T19:28:32Z

Ted.Szoczei:

{{EnglishNav}}

<div style="background:#98FB98;padding:5px;border:1px solid #006400;">
UNDER CONSTRUCTION
</div>
<p>Not a whole lot I can say: I don't know how Signalsoft is integrating this in to the simulations' UI.</p>
<p>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.</p>
<h3>Usage</h3>
<p>Requires multiplayer [[Line Operations Manual|Dispatcher Supervisor]]. Describe how Voice Communications appears in the simulation UI.</p>
<p>How game is affected?<br />
Ie. Change in buttons</p>
<h5>Calling a Neighbour</h5>
<p>Describe changes to operation of [[Communication|Communications panel]]<br />
Describe changes to operation of [[Line_Operations_Manual#Telephone|Telephone panel]]</p>
<h3>Port Forwarding</h3>
<p>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.</p>
<p>See [[Multiplayer_Manual#Ports|Multiplayer Ports and Forwarding]]</p>
<h3>VoIP</h3>
<p>What is VoIP?</p>
<p>Yes, it's standard RFC3261 SIP</p>
<p>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.</p>
<h3>Traffic</h3>
<p>All network traffic is direct, no intermediate servers are used.</p>
<p>Over the Internet all messages and media go directly from your router to the other player's router.</p>
<p>On LAN, all traffic goes directly from your machine to the other one, without going to the Internet.</p>
<h3>Network Bandwidth Usage</h3>
<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>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.</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>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.</p>
<p>A Normal mode stream requires bandwidth of 64 Kbps. An Enhanced mode stream requires 128 Kbps.</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>
<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}}

Voice Communications

2012-12-19T19:25:28Z

Ted.Szoczei:

{{EnglishNav}}

<div style="background:#98FB98;padding:5px;border:1px solid #006400;">
UNDER CONSTRUCTION
</div>
<p>Not a whole lot I can say: I don't know how Signalsoft is integrating this in to the simulations' UI.</p>
<p>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.</p>
<h3>Usage</h3>
<p>Requires multiplayer [[Line Operations Manual|Dispatcher Supervisor]]. Describe how Voice Communications appears in the simulation UI.</p>
<p>How game is affected?<br />
Ie. Change in buttons</p>
<h5>Calling a Neighbour</h5>
<p>Describe changes to operation of [[Communication|Communications panel]]<br />
Describe changes to operation of [[Line_Operations_Manual#Telephone|Telephone panel]]</p>
<h3>Port Forwarding</h3>
<p>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.</p>
<p>See [[Multiplayer_Manual#Ports|Multiplayer Ports and Forwarding]]</p>
<h3>VoIP</h3>
<p>What is VoIP?</p>
<p>Yes, it's standard RFC3261 SIP</p>
<p>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.</p>
<h3>Traffic</h3>
<p>All network traffic is direct, no intermediate servers are used.</p>
<p>Over the Internet all messages and media go directly from your router to the other player's router.</p>
<p>On LAN, all traffic goes directly from your machine to the other one, without going to the Internet.</p>
<h3>Network Bandwidth Usage</h3>
<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>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.</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>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>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>
<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}}

Voice Communications

2012-12-19T19:24:09Z

Ted.Szoczei:

{{EnglishNav}}

<div style="background:#98FB98;padding:5px;border:1px solid #006400;">
UNDER CONSTRUCTION
</div>
<p>Not a whole lot I can say: I don't know how Signalsoft is integrating this in to the simulations' UI.</p>
<p>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.</p>
<h3>Usage</h3>
<p>Requires multiplayer [[Line Operations Manual|Dispatcher Supervisor]]. Describe how Voice Communications appears in the simulation UI.</p>
<p>How game is affected?<br />
Ie. Change in buttons</p>
<h5>Calling a Neighbour</h5>
<p>Describe changes to operation of [[Communication|Communications panel]]<br />
Describe changes to operation of [[Line_Operations_Manual#Telephone|Telephone panel]]</p>
<h3>Port Forwarding</h3>
<p>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.</p>
<p>See [[Multiplayer_Manual#Ports|Multiplayer Ports and Forwarding]]</p>
<h3>VoIP</h3>
<p>What is VoIP?</p>
<p>Yes, it's standard RFC3261 SIP</p>
<p>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.</p>
<h3>Traffic</h3>
<p>All network traffic is direct, no intermediate servers are used.</p>
<p>Over the Internet all messages and media go directly from your router to the other player's router.</p>
<p>On LAN, all traffic goes directly from your machine to the other one, without going to the Internet.</p>
<h3>Network Bandwidth Usage</h3>
<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>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>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>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>
<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}}

Voice Communications

2012-12-19T18:51:31Z

Ted.Szoczei:

{{EnglishNav}}

<div style="background:#98FB98;padding:5px;border:1px solid #006400;">
UNDER CONSTRUCTION
</div>
<p>Not a whole lot I can say: I don't know how Signalsoft is integrating this in to the simulations' UI.</p>
<p>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.</p>
<h3>Usage</h3>
<p>Requires multiplayer [[Line Operations Manual|Dispatcher Supervisor]]. Describe how Voice Communications appears in the simulation UI.</p>
<p>How game is affected?<br />
Ie. Change in buttons</p>
<h5>Calling a Neighbour</h5>
<p>Describe changes to operation of [[Communication|Communications panel]]<br />
Describe changes to operation of [[Line_Operations_Manual#Telephone|Telephone panel]]</p>
<h3>Port Forwarding</h3>
<p>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.</p>
<p>See [[Multiplayer_Manual#Ports|Multiplayer Ports and Forwarding]]</p>
<h3>VoIP</h3>
<p>What is VoIP?</p>
<p>Yes, it's standard RFC3261 SIP</p>
<p>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.</p>
<h3>Traffic</h3>
<p>All network traffic is direct, no intermediate servers are used.</p>
<p>Over the Internet all messages and media go directly from your router to the other player's router.</p>
<p>On LAN, all traffic goes directly from your machine to the other one, without going to the Internet.</p>
<h3>Network Bandwidth Usage</h3>
<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>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>
<p>There is no traffic 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, 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>Since audio is continuous, Voice Communications sends the sa So limiter is upload: <br />
I could be using 1/2 my upload bandwidth while talking 128 Kbps</p>
<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>
{{EnglishNavBottom}}

Voice Communications

2012-12-19T18:03:31Z

Ted.Szoczei:

Voice Communications

2012-12-19T18:01:37Z

Ted.Szoczei:

Voice Communications

2012-12-19T17:54:38Z

Ted.Szoczei:

Voice Communications

2012-12-19T17:54:12Z

Ted.Szoczei:

Voice Communications

2012-12-19T17:52:07Z

Ted.Szoczei:

Voice Communications

2012-12-19T17:47:49Z

Ted.Szoczei:

Voice Communications

2012-12-19T17:42:07Z

Ted.Szoczei:

Voice Communications

2012-12-19T17:41:15Z

Ted.Szoczei:

Voice Communications

2012-12-19T17:39:39Z

Ted.Szoczei:

Voice Communications

2012-12-19T17:33:04Z

Ted.Szoczei:

Voice Communications

2012-12-19T17:29:20Z

Ted.Szoczei:

Voice Communications

2012-12-19T17:26:48Z

Ted.Szoczei:

Voice Communications

2012-12-19T17:09:22Z

Ted.Szoczei:

{{EnglishNav}}

<div style="background:#98FB98;padding:5px;border:1px solid #006400;">
UNDER CONSTRUCTION
</div>
<p>Not a whole lot I can say: I don't know how they're integrating</p>
<p>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.</p>
<h3>Integration?</h3>
<p>Requires multiplayer (add link)</p>
<h3>Ports</h3>
<p>Extended range</p>
<p>(put link to multiplayer)</p>
<p>Forwarding</p>
<h3>Traffic</h3>
<p>All traffic is direct, no servers are used</p>
<p>Over the Internet all messages and media go direct from your router to the other router</p>
<p>On LAN, all traffic goes directly from your machine to the other one, without going to the Internet</p>
<h3>Updates</h3>
<p>To user, updates are automatic but should be prompted</p>
<p>Before loading VCP component, Sim should call updater, updater can look for update and return value if new available</p>
<p>Then sim should prompt user to install update</p>
<p>If user accepts then sim call updater to download and replace</p>
<h3>VoIP</h3>
<p>What is VoIP?</p>
<p>Yes, it's standard RFC3261 SIP</p>
<p>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.</p>
<h3>Bandwidth used</h3>
<p>Traffic goes both ways</p>
<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>LANs are 10 or 100 Mbps<br />
WiFi is 3Mbps<br />
My DSL internet is 4.6 to 5.3 Mbps down and 310 to 740 Kbps up<br />
So limiter is upload: <br />
I could be using 1/2 my upload bandwidth while talking 128 Kbps</p>
<p>LAN traffic is direct, not through router</p>
<h3>Usage</h3>
<p>Communications panel?<br />
Calling a Neighbour</p>
<p>Telephone panel?</p>
<h3>Logic</h3>
<p>Ie. How game is affected<br />
Change in buttons</p>
{{EnglishNavBottom}}

Voice Communications

2012-12-19T17:04:00Z

Ted.Szoczei:

{{EnglishNav}}

<div style="background:#98FB98;padding:5px;border:1px solid #006400;">
UNDER CONSTRUCTION
</div>
<p>Not a whole lot I can say: I don't know how they're integrating</p>
<p>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.</p>
<h3>Integration?</h3>
<p>Requires multiplayer (add link)</p>
<h3>Ports</h3>
<p>Extended range</p>
<p>(put link to multiplayer)</p>
<p>Forwarding</p>
<h3>Traffic</h3>
<p>All traffic is direct, no servers are used</p>
<p>Over the Internet all messages and media go direct from your router to the other router</p>
<p>On LAN, all traffic goes directly from your machine to the other one, without going to the Internet</p>
<h3>Updates</h3>
<p>To user, updates are automatic but should be prompted</p>
<p>Before loading VCP component, Sim should call updater, updater can look for update and return value if new available</p>
<p>Then sim should prompt user to install update</p>
<p>If user accepts then sim call updater to download and replace</p>
<h3>VoIP</h3>
<p>What is VoIP?</p>
<p>Yes, it's standard RFC3261 SIP</p>
<p>No registration is needed<br />
Address lookup is provided by signalsoft multiplayer server</p>
<h3>Bandwidth used</h3>
<p>Traffic goes both ways</p>
<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>LANs are 10 or 100 Mbps<br />
WiFi is 3Mbps<br />
My DSL internet is 4.6 to 5.3 Mbps down and 310 to 740 Kbps up<br />
So limiter is upload: <br />
I could be using 1/2 my upload bandwidth while talking 128 Kbps</p>
<p>LAN traffic is direct, not through router</p>
<h3>Usage</h3>
<p>Communications panel?<br />
Calling a Neighbour</p>
<p>Telephone panel?</p>
<h3>Logic</h3>
<p>Ie. How game is affected<br />
Change in buttons</p>
{{EnglishNavBottom}}

Voice Communications

2012-12-19T17:03:15Z

Ted.Szoczei:

{{EnglishNav}}

<div style="background:#98FB98;padding:5px;border:1px solid #006400;">
UNDER CONSTRUCTION
</div>
<p>Not a whole lot I can say: I don't know how they're integrating</p>
<p>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.</p>
<h3>Integration?</h3>
<p>Requires multiplayer (add link)</p>
<h3>Ports</h3>
<p>Extended range</p>
<p>(put link to multiplayer)</p>
<p>Forwarding</p>
<h3>Traffic</h3>
<p>All traffic is direct, no servers are used</p>
<p>Over the Internet all messages and media go direct from your router to the other router</p>
<p>On LAN, all traffic goes directly from your machine to the other one, without going to the Internet</p>
<h3>Updates</h3>
<p>To user, updates are automatic but should be prompted</p>
<p>Before loading VCP component, Sim should call updater, updater can look for update and return value if new available</p>
<p>Then sim should prompt user to install update</p>
<p>If user accepts then sim call updater to download and replace</p>
<h3>VoIP</h3>
<p>What is VoIP?</p>
<p>Yes, it's standard RFC3261 SIP</p>
<p>No registration is needed<br />
Address lookup is provided by signalsoft multiplayer server</p>
<h3>Bandwidth used</h3>
<p>Traffic goes both ways</p>
<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>LANs are 10 or 100 Mbps<br />
WiFi is 3Mbps<br />
My DSL internet is 4.6 to 5.3 Mbps down and 310 to 740 Kbps up<br />
So limiter is upload: <br />
I could be using 1/2 my upload bandwidth while talking 128 Kbps</p>
<p>LAN traffic is direct, not through router</p>
<h3>Usage</h3>
<p>Communications panel?<br />
Calling a Neighbour</p>
<p>Telephone panel?</p>
<h3>Logic</h3>
<p>Ie. How game is affected<br />
Change in buttons</p>
{{EnglishNavBottom}}

Voice Communications

2012-12-19T16:46:47Z

Ted.Szoczei: Initial draft

{{EnglishNav}}

<div style="background:#98FB98;padding:5px;border:1px solid #006400;">
UNDER CONSTRUCTION
</div>
<p>Not a whole lot I can say: I don't know how they're integrating</p>
<p>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.</p>
<h3>Integration?</h3>
<p>Requires multiplayer (add link)</p>
<h3>Ports</h3>
<p>Extended range</p>
<p>(put link to multiplayer)</p>
<p>Forwarding</p>
<h3>Traffic</h3>
<p>All traffic is direct, no servers are used</p>
<p>Over the Internet all messages and media go direct from your router to the other router</p>
<p>On LAN, all traffic goes directly from your machine to the other one, without going to the Internet</p>
<h3>Updates</h3>
<p>To user, updates are automatic but should be prompted</p>
<p>Before loading VCP component, Sim should call updater, updater can look for update and return value if new available</p>
<p>Then sim should prompt user to install update</p>
<p>If user accepts then sim call updater to download and replace</p>
<h3>VoIP</h3>
<p>What is VoIP?</p>
<p>Yes, it's standard RFC3261 SIP</p>
<p>No registration is needed<br />
Address lookup is provided by signalsoft multiplayer server</p>
<h3>Bandwidth used</h3>
<p>Traffic goes both ways</p>
<p>No traffic while not talking?</p>
<p>Normal and (name?) voice: 64 Kbps, 128 Kbps<br />
LANs are 10 or 100 Mbps<br />
WiFi is 3Mbps<br />
My DSL internet is 4.6 to 5.3 Mbps down and 310 to 740 Kbps up<br />
So limiter is upload: <br />
I could be using 1/2 my upload bandwidth while talking 128 Kbps</p>
<p>LAN traffic is direct, not through router</p>
<h3>Usage</h3>
<p>Communications panel?<br />
Calling a Neighbour</p>
<p>Telephone panel?</p>
<h3>Logic</h3>
<p>Ie. How game is affected<br />
Change in buttons</p>
{{EnglishNavBottom}}