AB7CQ & N7RPT

email: ab7cq@ebidpal.com

Tombstone, AZ 85638

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WIRES-X Help

 

Item 1:Yaesu System Fusion Sites

Item 2: WIRES-X Settings

Item 3: Mic Gain - Speech Processing

 

Item 1:Yaesu System Fusion Sites

 

Posted by John Kruk, N9UPC, National Sales Manager, Amateur Division, Yaesu USA in Yaesu System Fusion Official Group

Time to clear up some confusion that has been popping up on some (Anti) Yaesu System Fusion sites. This is in regards to YSF and FCS reflectors.

 

The first one and the biggest one is that YSF and FCS reflectors are not owned, maintained, designed, or monitored by Yaesu in any way, shape, or form. Yaesu System Fusion (C4FM) is an open protocol and people have made these types of servers or reflectors for Yaesu System Fusion voice protocol.

 

Some have crossed these FCS and YSF reflectors over to the WiRES-X network but they are NOT, and I stress NOT, a direct connection to the WiRES-X network. The only thing that can be directly connected to the WiRES-X network is a HRI-200.

 

These devices have caused some audio issues especially when doing a crossmode set-up (one digital voice mode to C4FM). This can create audio issues or issues in where people are not heard but the callsign appears. This is NOT a WiRES-X issue but usually a crossmode issue with the device.

 

Remember that these devices that give you access to these reflectors are not true Yaesu System Fusion (C4FM) transmitters as they do NOT have a DP-ID. Therefore, you need to be careful if you choose to block a node or room because of the traffic or issues created by these bridges.

 

EXAMPLE: If you have a crossover from a device using an FCS or YSF reflector to a WiRES-X node the node ID will be shown across the room. Therefore, if you block that node because of issues you take out everything with that node. Be it the users with Yaesu System Fusion radios going into the node directly or the devices carrying the FCS or YSF reflector traffic.

 

WiRES-X nodes or rooms use actual numbers as the address to the room. Therefore, when you attempt to save a room into a WiRES-X memory on a radio they will save. If you attempt to save an FCS or YSF reflector it WILL NOT work. EXAMPLE: Yaesu nodes start with a 1xxxx or a 3xxxx number and rooms start with a 2xxxx or a 4xxxx number. So if you want to store say room 21001 into a WiRES-X memeory slot you can enter 21001 or the room name. YSF reflectors do not have a number associated with them and FCS numbers do not work. Even with FCS the server 00x and reflector xx equal a five digital number the number would always stat with a 0 (like 00xxx). The radio knows that 0 is not a valid number and will not save it. It might until the radio powers off but then it will be gone.

 

This is just to clear up questions about FCS and YSF that users have had recently. This is not a post about saying which one is better than the other. WiRES-X operation is very different then FCS and YSF operation. Thank you for reading this and I hope that this answers peoples questions.

 

Item #2: WIRES-X Settings

 

In order to make using WIRES-X a more pleasant experience I would like to share a few suggestions regarding how to setup WIRES-X.

 

One of the most important adjustments is setting up your transmit audio level (mic gain)to avoid audio over modulation and clipping. Quite often, while listening to WIRES-X users engaged in a conversation, either one, or both, comment that their audio is choppy, or being cutoff, some don't just don't mention it to each other.

 

There are two principle reasons for this problem, the first reason is that users fail to properly adjust their transmit audio level within the WIRES-X software and overdrive their transceiver transmitters which results in audio clipping, audio clipping can be repeated through the local repeater, or on simplex transmissions, fact is, it's just damn annoying to everyone else listening.

 

The second problem has to do with incomplete "audio packets. "Quite often when listening to conversations in progress on WIRES-X users will comment that part of the their conversation was "chopped off" at the beginning or end of their comments, once again, some don't just don't mention it to each other.

 

What most users do not take into account is that there's a "lag time" for the audio packets to travel from the user's node station through the Internet network, and as I understand the Yaesu Server has to validate each node, then to the user's receiving node/repeater station. The user must press their PTT and allow a few moments before beginning their transmission and allow a few extra moments at the end of their transmission before releasing the PTT, this allows all the hardware along the Internet path to lock up and for the audio packets to complete the trip from node to node.

 

Another concern has to do with your transceiver microphone, they're not all created equal, some mic's are hotter, or softer than others, a transceiver with a hot mic will cause audio clipping, some users simply speak louder into the mic while others speak more softly into the mic. Point here is, you should set your microphone transmit audio and receive audio levels using the built in WIRES-X software controls.

 

If your transceiver is equipped with an audio control adjustment (mic gain control), or a speech compressor, you will also need to adjust these controls in addition to the the audio adjustment controls in the WIRES-X PC software.

 

Some years ago ICOM equipped a lot of their VHF/UHF transceivers with microphones that were way too hot, in fact ICOM actually offered a replacement chip resistor for the microphone to pad down the audio. The down side was that unless you possessed the equipment and skills to work with surface mounted componets, many were not able to make the repair without damaging the componets.

 

Quite often operators, especially new Hams will set the mic gain too high and drive up the speech compression, which either causes the audio to clip out of the pass band, or the audio generates mostly low tones with all bass and becomes un-understandable. This can also happen if a mic is too HOT, then one needs to back off the mic, or modify the mic. This problem can be accentuated when going through a repeater.

 

Also remember if you hold your microphone too close to your mouth, there's a good chance that you'll be over driving your transceiver transmitter, your audio will be clipped/distored/muffled.

 

Lastly, if your Internet connection lacks sufficient speed, you can experience audio packet dropouts. This can also happen if you have networked numerous devices in your home/shack and it's being overloaded. If your involved with running multple computers, WIRES-X, IRLP, an IMRS LAN, wireless phones, tablets, a networked weather station, Internet gaming, TV's and other VOIP communications, all being pumped through your router, it's best that you contract with an Internet provider that can provide you with a "true static IP" (connection), not dynamic, and sufficient speed to handle all the data/devices you are, or plan to have operational in your home/shack. A router can become a real choke point.

 

Generally speaking, you should use the necessary equipment to adjust your audio levels, problem is, most do not own lab instruments for this purpose, equipment like spectrum analyzers, oscilloscopes, tone generators, and other lab test gear to perform these adjustments, in that case, you might be able to request help from an Elmer, club friend, or other ham skilled in performing these adjustments, most are willing to help if you ask. With time and experience you will develop a feel for adjusting these settings without the need of test instruments, however, the best way is to use appropriate test gear for making these adjustments.

 

There are temporary short term shortcuts to adjusting your audio level(s) without employing test equipment, connect your transceiver to a dummy load, then listen to your transmissions on a secondary transceiver, or receiver monitor and adjust your transmit audio level accordingly.

 

Now Let's Discuss Sending DTMF/Digital Tone Codes)

 

Most software, like WIRES-X allows one to adjust audio levels, additionally, many modern VHF/UHF/HF transceivers now allow one to adjust microphone audio/speech compression audio. If these controls are not set properly you will be over driving your transmitter, which in turn can cause "clipping", exceed deviation specifications causing your tones not to be recognized/decoded, and cause spurious products to be produced. There are deviation limits that must be observed.

 

Many transceivers like the Yaesu FTM-100D, FTM-400XDR and many others allow one to adjust the sensitivity of the microphone (gain), these levels can be adjusted to minimum, low, normal, high, & maximum and you can easily overdrive your transmitter if you speak too loudly into the microphone.

 

So how do I check, or make these adjustments. Once again you will need to use the appropriate test equipment, a tone generator, spectrum analyzer, oscilloscope, etc., and of course, you will need the assistance of someone skilled to make these adjustments.

 

Once again, there are short term shortcuts to adjusting your audio level(s) without employing test equipment, connect your transceiver to a dummy load, then listen to your transmissions on a secondary transceiver, or receiver monitor and adjust your transmit audio level accordingly.

 

Timing of sending tones is also important, become acquainted with your microphone and execute the tones(s) in a uniform manner, if you don't, you may well encounter problems with the receiving station to properly decode your tones.

 

Once again, you can request help from an Elmer, club friend, or other ham skilled in performing these adjustments, most are willing to help if you ask. That's what ham radio is all about, helping each other so we can share information and learn from each other.

 

None of this subject matter is intended to scare the new ham, it's intended as an aid so ham radio will be an enjoyable experience for everyone. None of us are the sharpest knife in the drawer, we all learn from each other, I've been learning everyday of my amateur radio life for 38 years and I've made my share of mistakes. 

 

This is the stuff they don't teach in most amateur radio license classes, just rote memorization. Amateur radio is not "plug & play, it requires motivation to learning and improving operating skills.

 

*************************

 

Technical Article #1 - Mic Gain, Speech Processing (Compression), and IMD Intermodular Distortion

 

Here's a little tutorial on mic gain and speech compression that expains how it all works in a transceiver.

 

New generation multiband multimode rigs typically cover VHF/UHF/HF, these type of rigs have independent mic gain and speech compression controls that the user can adjust independently, older transceivers like a typical mobile rig do not allow you to adjust mic gain or speech compression, it’s all preset in older VHF/UHF mobile rigs.

 

Quite often operators, especially new Hams will set the mic gain too high and drive up the speech compression, which either causes the audio to clip out of the pass band, or the audio generates mostly low tones with all bass and becomes un-understandable. This can also happen if a mic is too HOT, then one needs to back off the mic, or modify the mic. This problem can be accentuated when going through a repeater.

 

The use of speech processing (aka speech compression or clipping) has become the major bane of amateur radio, especially mobile operation. It allows every little nuance of engine noise, AC fans, the kids in the back seat, and that squeak in the left quarter panel to be plainly heard. It is important to remember, the average vehicle traveling at 60 mph, is at least 25 dB louder than the average living room, and some are over 40 dB louder.

 

Most amateurs do not know how to properly set their microphone and/or processing controls, resulting in some really lousy-sounding, on-air signals. They might sound passable when properly tuned in, but a few kHz away the distortion products (IMD Intermodular Distortion) can be clearly heard.


It should be noted at this point, that any form of speech processing does increase the average power level. This fact does cause the signal to appear louder on the receiving end. However, if used excessively, it also removes most of the nuances our brain uses to comprehend what's being said.

Almost without exception, all modern HF transceivers have some form of speech processing built-in (unfortunately!). Speech compression in its simplest form, is nothing more than an automatic level control. The softer nuances of speech are amplified more than the louder ones. In some cases, a different (narrower) bandpass filter is used and/or different DSP settings, and the dynamic nature of the ALC may be changed to minimize IMD. However, over zealous use (excessive gain and compression levels) will defeat any provision to prevent excessive IMD!

Speech processing, however it is done, not only increases the average output power, is also increases the average current draw! Depending on the configuration, it could in fact double! This can easily tax the heartiest of electrical systems, especially when running high power. As the voltage sags, the IMD goes up, and readability decreases even more.

Here is the very best advice you'll ever get with respect to using speech processing and/or excessive microphone gain in a mobile—don't use it!

 

The problem is that most new hams are excited with their new rig and their focus is on the new toy, they fail to take the time to understand how the transceiver really works and how these features affect the tranceivers overall peformance.

 

Unfortunately, a lot of this is overlooked in amateur radio classes. Misadjustment of these controls can cause all sorts of negative issues, one of which is generating spurious emissions on the bands, something extremely frowned on by hams, their signals spread over a wider specturm of frequency (called buck shotting), they can even be cited by the FCC for generating spurious transmissions and even fined (issued a forfeiture) for generating spurious emissions.

 

When you're running the WIRES-X PC software look at the menu bar and you'll see an option titled "VIEW", click on it, cursor down until you see Audio Adjustment(s) and select it, this is where you can make your audio adjustments, I would suggest that you monitor your transmit audio utilizing a dummy load connected to your rig during these adjustments so you're not responsible for causing interference on the frequency, if you have the necessary test gear, you can monitor your audio on a spectrum analyzer, oscilloscope, and use a tone generator, perhaps a ham friend/Elmer can help you with this, there are even some free PC software based audio analyzers that you can use for this purpose.

 

Please keep in mind that the Internet path between stations can be responsible for communications audio packet dropouts, not all stations use the same path, Internet communication paths can be perfect to marginal, also the speed of your hosting provider plays a significant role in Internet communications. By all means monitor your WIRES-X Node/Repeater, you're responsible for any interference you cause, DO NOT LEAVE YOUR WIRES-X NODE UNATTENDED, be responsible and assign someone as a control operator if you plan to be away from your repeater/node so they can disable it until the issue can be resolved.

 

Bob Krueger, AB7CQ

Web Administrator

RPTR 1: 146.920/146.320 PL 123.0 (WIRES-X Room 28187)

RPTR 2: 444.600 / 449.600 PL 100  (LAN Linked)

Simplex IRLP Node: 7515 PL114.8 (146.540)

Email: ab7cqradio@ebidpal.com


 


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