Recently in Amateur Radio Category

Callsigns in IPv6

http://mesa-arc.org/operating/IPv6 reports that:

Embedded Callsigns in IPv6 Addresses

Club Members Jacques N1ZZH and Vinnie N1LQJ have developed a method of embedding a 2x5 (7 Character) callsign plus up to 185 nodes, plus 1 universal bit and three reserved bits in the 2nd octet, and a 16 bit amateur radio identifer at bit 24 of an IPv6 /64 Subnet address. 

Sample programs and information on this proposed standard can be found at http://sourceforge.net/projects/hamv6



NICE!

GMSK Tutorial and Implementation

From Spread Spectrum Scene, http://sss-mag.com/index.html (go read everything they have!), there's an app-note from MX about GMSK entitled: Practical GMSK Data Transmission.

Here's the link from the 'scene: http://sss-mag.com/pdf/gmsk_tut.pdf and a local link.

This PDF explains and shows the phase-discontinuous nature of PSK and other modulations:  http://www.emc.york.ac.uk/reports/linkpcp/appD.pdf
 
What people don't immediately realize is that GMSK is MSK, with rounded edges. Why is that important? Because square waves are by definition made up of the fundamental frequency and all of the odd harmonics of that frequency. However, when you cram a square wave into a multiplication process, like modulation, it creates mixing products -- lots of them. We call these products "sidebands" and typically try to reduce them through filters in some way.

Filtering can be done before or after modulation. It is more effective, and a better use of power to filter before the modulation process than after. In the case of GMSK, a specially designed Gaussian filter is used, which has two effects on the signal. One effect the Gaussian filter has is that it is intentionally set lower than the frequency passing through it. This causes it to have a rather pronounced effect on the signal passing through; specifically on the Gaussian filter, this gives rise to "InterSymbol Interference" or ISI. This interference happens because the filter causes the bits to lag a little in the time domain, causing the receiving modem to occasionally decide a bit is a one when it should be a zero or a zero when it should be a one. The second effect of the Gaussian filter is that it turns the sharp, square edges of the data into a rounded spike-like pulse. This rounded pulse has advantages.

The rounded Gaussian pulse has many advantages. Recall that in FM radio, a modulating signal is applied to a modulator. As the signal sweeps in one direction, say from zero to V+, the resulting frequency of the transmitter rises to a set maximum. Likewise, as the signal sweeps from zero to V-, the transmitter lowers in frequency to another set minimum. The advantage to the rounded spike is that, much like a plotter connected to an oscilloscope, the path is easily followed with smooth transitions. This is extremely important, as sudden jumps in amplitude and frequency cause phase discontinuities which make the signal hard to process. In Frequency Modulation, the signal must always remain phase-continuous without rapid reversals. The reason for this is manyfold, but I'll settle on two: 1) If it wasn't, then it isn't by definition FM, and 2) because the most common amplifier used to amplify FM signals, the often 80% efficient Class C, requires that the signal "ring" or spend half the time per cycle coming out of a RC "tank". So the Gaussian pulse allows the modulator to follow along, smoothly, generating a signal with minimal harmonics as a result of the modulation process. Again, because of this smoothness of action, rather than jumping around like QPSK or QAM, the GMSK signal is FM-complaint, allowing common, off the shelf gear to be used to generate, receive and amplify the GMSK signal. In short, GMSK appears to be a preferred way to transmit FSK signals, mostly due to removing unwanted sidebands which would otherwise be wasted as heat in the filters following the modulator. Bt=0.3 seems best, apparently the commercial data guys are aiming for 0.25 and 0.27.

Remember, of course, that MSK is defined as the shift at half the frequency of the data, and the modulation index is 0.5. GMSK doesn't get all the way down there, but it's still spectrally more efficient than FSK. Often, the same gear can receive GMSK as FSK, particularly the MX589 used by Kantronics in the KPC9612.

From the commercial perspective, I want to point out a few things:

1) Motorola has for years, since the Syntor X series, provided a split low-frequency and high frequency modulation path. This allows the CTCSS tones to be modulated at the lower frequencies that the VCO's PLL would otherwise automatically remove as detected frequency error.

2) Motorola has also used multi-level FSK to transmit and receive trunking control information.

FSK has the advantage of simple transmission -- attach the data stream to the modulator -- and simple detection: one need only use a data slicer to recover the data if the center frequency or one of the frequencies is known. 

Seeing as how the Motorola Spectra, Saber, and X9000 lines are based heavily off of the 68HC11, and the Kantronics KPC-9612, -9612+, -3, and -3+ all use the HC11 as well, I would not be surprised if Motorola was bit-banging data in from the trunking control channel without a modem much like how Kantronics recovers data in the 1200 BPS port of the above mentioned TNCs.

Update: I've just become aware that GMSK is being referred to by some as "GFSK with a modulation index of 0.5." GFSK is "Gaussian Frequency Shift Keying", per Analog Devices ADF7023-J product information: http://www.analog.com/static/imported-files/data_sheets/ADF7023-J.pdf

Motorola Mitrek and TinyTrak3

I can't believe this hasn't come up, but a little searching seems to indicate that not very many folks interface a Motorola Mitrek to a TinyTrak3.

I ran into an issue with a Motorola Mitrek radio and a TinyTrak3. I was trying to determine what the radio was expecting as far as a modulating signal was concerned. Motorola, in the lowband Mitrek manual, indicates that a 1KHz tone should be generated at 1V peak-to-peak and feed into the microphone input through a .33uF capacitor. The DEV adjustment should be turned until deviation reaches 4.8KHz, and PL deviation should be between 0.5 and 1.0 KHz.

Short story since I'm light on time. Remove/don't install R8 into the TinyTrak3. This is used to couple PTT to TX Audio or vice versa. It simulates how HT external microphones used to be wired (2.2Kohms to ground to transmit). Short R5. I tried halving the resistance with another 220Kohm resistor for 105Kohm... Just nix that altogether and short R5. This can be done with a solder bridge, leaving you the option of going back to 220Kohm later if you need to.

I repeat: don't install R8 or remove R8 and short R5.

Short version again: Short R5 on the TinyTrak3 if you need more drive. See page 15 of the TinyTrak3Plus manual: "No audio is heard on a receiver." See page 14 of the TinyTrak3 manual under the same heading. It's the same paragraph verbatim.

Then set the  R6 to maximum. Tell the TinyTrak3 to generate both tones and adjust the IDC pot on the channel element for -/+5.0KHz deviation. Then adjust R6 for 4.0KHz deviation.

(The reason for doing this is that when the IDC (limiter) kicks in, it introduces clipping, distortion, and harmonics to the signal. By setting the channel element for 5KHz, the limiter is high enough to prevent introducing distortion through normal peak clipping. The stock Mitrek microphone circuits contain a brick-wall splatter filter to prevent excessive modulation/sidebands/harmonics.)

This message is my basis for that assumption of 4.0KHz deviation:
http://groups.yahoo.com/group/MicroTrak/message/844

Reproduced here:

From: WA8LMF

Michael Crowder wrote:
> Rudy, Thanks for the assistance.
>
> I've tried re-loading a new config, but I still have the problem that
> the 1200Hz tone is twice the amplitude of the 2200Hz tone. I'm using
> the TinyTrak3 configuration program to cause the MicroTrak8000 to send
> the separate tones, and then looking at the received signal on my PC
> using AGWPacket engine. Anyone have any ideas what may be causing the
> different levels in tones?
>
> --
> M


This is perfectly normal if you are taking the audio off the speaker
terminal of the receiver.

You are seeing the normal de-emphasis curve of a voice FM receiver. At
the transmit end, audio fed into the mic jack has the high frequencies
boosted relative to the low. Ideally, this boost or pre-emphasis should
be 6dB/octave over the range from 300 to 3000 Hz. The over-the-air
deviation or level of the tone is directly proportional to it's
frequency; i.e. a tone at 2400 Hz should have twice the deviation as one
at 1200 Hz. At the receiver, a corresponding de-emphasis network
(high-frequency cut) cancels out the boost applied at the sending end
for a net flat (equal levels for all audio frequencies) response at the
speaker.

However, if you apply the TX audio to the 6-pin mini-DIN "data" or
"packet jack" (on radios that have it) instead of the mic jack, no
transmit pre-emphasis is applied -- both tones go out at exactly the
same level. [Note that the MicroTrak integrated TT3 and transmitter has
no mic input with pre-emphasis - the TT is coupled directly to the TX
modulator with net flat response.] At the RX end, you continue to
apply the high-freq roll-off (de-emphasis) [assuming you are taking the
RX audio off the speaker or equivalent] with the result that the
attenuated 2200 Hz high tone is now about HALF the amplitude of the 1200
Hz low tone.

Historically, 1200 baud packet has been a "jam it in the mic jack of any
radio" (with the corresponding pre-emphasis applied) convention, while
9600 baud packet has been coupled directly into the transmit modulator
(with the resulting flat response). At the receive end, 1200 baud has
customarily been taken off the (de-emphasized) speaker output while 9600
receive has to be taken directly off the receiver discriminator (flat
response) before any high-freq de-emphasis is applied.

Traditionally, 1200 baud packet devices have been set to yield about
3.5-4.0 KHz deviation on the high tone which yields about 2.0-2.5 KHz
deviation on the low tone. Devices such as the Kenwood APRS radios and
the Microtraks that transmit "flat" response at 1200 baud are in
conflict with decades of packet convention that says 1200 baud
transmissions should be pre-emphasized.

[
Many hardware-based TNCs are very intolerant of unequal tone levels,
especially when the high tone is LOWER in level than the low tone. The
AGW Packet Engine and MixW software modems, on the other hand, are quite
tolerant of mis-matched tone levels. The result is that you may be able
to successfully decode your own transmissions with skewed tone levels in
your soft TNC setup, while a digipeater equippped with a TNC2 or KPC3
hardware TNC connected to the (de-emphasiszed) speaker out of it's radio
will not.

The solution is to apply pre-emphasis to the transmitted signal. Place
a small capacitor (probably somewhere between .01 and .005 uf -- you
will have to experiment) in series between the TinyTrak TX audio out and
transmitter audio-in. If the cap is small enough, it will attenuate the
low frequency relative to the high frequency enough to create a net
pre-emphasis effect. In your RX setup with de-emphasis, you should then
observe that the two tone levels are now nearly the same.


--

Stephen H. Smith wa8lmf (at) aol.com
EchoLink Node: 14400 [Think bottom of the 2M band]
Home Page: http://wa8lmf.com --OR-- http://wa8lmf.net

The service monitor verifies these levels, total deviation around 4.0KHz (below the IDC limit) for both tones or the high tone, and around 2.5KHz deviation for the low tone alone.

Another suggestion from Bruce Lane via The Batboard is to set the 40W radio to 15W. This appears to limit current draw to about 6A at 13.8V. Seeing as how Motorola said not less than 55W on the 100W Syntors, I'm guessing it's not a good idea to go any lower on the Mitrek lest the PA become unstable and transmit spurs:

http://batboard.batlabs.com/viewtopic.php?f=1&t=13319&start=0

Youbetcha. I've had excellent results from a modified Mitrek. The mods involved were changing two capacitors in the transmitter section (so the multipliers would tune up -- I had the 150.8-174 split radio), and having International Crystal re-do the channel elements for 144.3900.

The only other change I made was to turn down the output power to about 15 watts (40-watt radio). Not only will this make the PA last just about forever, it also cuts down the amount of current needed on transmit.

If you go with a Mitrek, let me know. I can help with a step-by-step.

73 de KC7GR

Repeater-Builder website is a great central resource:
http://www.repeater-builder.com/mitrek/mitrek-index.html

SEITS has some info as well:
http://www.seits.org/repeater/mitrek/mitrek.htm

SRGClub has a great write up with some images you might need if you're like me and don't have the service manual in front of you.

http://www.srgclub.org/images/Mi_YesCR903-Schem.jpg
http://www.srgclub.org/images/Mi_YesCR903-Layout.jpg
http://www.srgclub.org/images/Mi_noCR903-Schem.jpg
http://www.srgclub.org/images/Mi_noCR903-Layout.jpg

R909 is TX Power, R927 is TX Limit.

Get a good pair of power cables, start turning the power up until you get to about 2W above rated. Then turn down the operating level. On mine, Limit is the blue pot, Operating Level is the red one.

I shouldn't have done it, but I cranked them both wide open. I got 58.3W out of a 40W rated radio. I didn't leave it that way -- I set it to 15W, then noted that I'm losing about a watt and some change in a RG-174 jumper. The service monitor only reported 13.3W.

The power supply was providing 13.73VDC, and because I was doing a power calibration, I hooked the radio up with a #2 welding cable and some half-inch ground braid I'd recovered and had handy. Losses inside the radio made the voltage at the back of the interface connector at 13.55VDC at 15W, and 13.45VDC at 45W.

[And then I found out the RG-174 jumper was bad and replaced it. Read more below...]

Those pictures are from this excellent write up:

http://www.srgclub.org/Mitrek_Rp_conversion.html

Here's another:

http://kf8kk.com/wi0ok/mitrek-conv1.htm

Connector pinouts:
http://www.n1uec.org/n1uec/mitrekrepeater.html

How I mocked up a control head:
http://www.repeater-builder.com/mitrek/mitrek-interfacing.html#Carrier_Squelch

Again, I suggest a service manual, but if you don't have one and you've seen one in the past, you might be able to wing it. 73 and good luck...

TinyTrak3 Manuals:
http://blog.catonic.us/kirby/tinytrak/TinyTrak3.pdf
http://blog.catonic.us/kirby/tinytrak/TinyTrak3Plus.pdf

TX Delay Settings:
http://info.aprs.net/index.php?title=NorCal_APRS_Regional_Info

My actual info on TXD comes from listening to the audio unsquelched and squelched while transmitting packets. The Mitrek is up and on the air in about 60ms. Since it's a crystal-based radio, it's on-channel as well -- no warm up needed. So a TXD of 8 or 80ms is long enough for the radio to send about 20 ms of preambling.

At 9600 BPS, these radios can be backed off to TXD 6 or 7 and used at speed.

The venerable Alinco DR-1200 needed a TXD of 33, mostly due to the broad and sweeping behavior of the VCO. That thing transmits Chirp FM spread-spectrum before it gets on frequency! It's at power before the VCO stabilizes. I'm surprised the FCC let them get away with it.

Most radios should be able to do a TXD of 33. If you listen and notice that the first part of the signal sounds like two alternating notes -- it is. That's the preamble. In practice, you want that to be as short as possible, while still decoding packets correctly. Much of that depends on the other radio returning from transmit to receive quickly. This is why the DIGIs now transmit immediately after hearing a packet. Because if you sent it, you're still waiting to swap back from transmit to receive, and the digi is taking advantage of the pause to blurt the message out.

I set the TinyTrak to a TXD of 18, as shown in the link above. The issue, again, is not the Mitrek, it's every other radio out there. Even with the reed relays, the Mitrek is a fast piece of 1970s technology.

SSID settings:
http://aprs.org/aprs11/SSIDs.txt
http://zlhams.wikidot.com/aprs-ssidguide
http://info.aprs.net/index.php?title=SymbolsAndSSIDs
http://vk3.aprs.net.au/aprs_ssid_usage.htm

In my case, I went with KE4AHR-12, ("one-way trackers"). Since it vehicle-borne, it will change to -9 sooner or later, and my bicycle will be -12 or some other SSID.

TinyTrak generic:
http://www.nwaprs.info/ttconfiguration.htm

03/28/12 Edit:

Per N8DEU, I backed the deviation off to 3.5KHz. I found the RG-174 jumper in the case was bad, and replacing it with another pre-made one caused the power out to jump from 13W to 18W with no change in current. After retuning the Mitrek to 25W (out of the box) the current draw was about 6.25A at 13.6V. The cigarette lighter is fused at 10A, so this gives me a few amps for operating another radio and charging a few mobile devices.

As deviation goes down, power efficiency goes up, however gains are on the order of 3dB, if that. You need at least 6dB to notice the difference.

I use a 1/4-wave antenna for my tracker. Its relatively small for the band used, and gives good performance out to certain angles. My perspective at this time is more coverage-testing of the existing nodes, and the dipole allows me to communicate using knife-edge diffraction. Now if only we'd implemented the system using 9600 BPS instead of 1200 BPS....

On the receive side, I mocked up a control head using a 10K ohm resistor between "Detected Audio" and "Squelch Wiper". A 3.3K ohm resistor connects "Squelch wiper" to audio ground. This gives a fixed squelch response. Then I connected the + side of the upper audio capacitor on the interconnect board to the backside of JU-3C, which connects to L18 -- the third large pin in the Mitrek control cable interface connector. From there, I routed the signal into the Signal Present line on the TinyTrak and set the voltage at about 3V. Reprogram the TinyTrak to invert DCD, and then set the bias so that it operates in a reasonable fashion. Net result is that the reciever works correctly when there is noise at -124dBm, or when there is a signal at -119dBm. The receiver actually starts hearing around -130dBm.

PE-75 Generator Manual TM 11-900

Original Post: 02-20-2012 @ 1:17:40

PE-75 Power Unit

2.5KW Generator, 120V, 22.5A, 1800 RPM
6.5HP Briggs & Stratton Model ZZ (with 'antique' updraft carburetor), 2400 RPM

330 lbs, mounted to steel frame.

Yes, I have a manual. Here's Part I, V1.0 of it:

[removed]

This is also known as TM 11-900.


This monster was a part of an SCR-197.

Here's some other manuals I found on the web. Mine, as you note, is slightly different:

http://blog.catonic.us/kirby/PE-75/

Edit: 6/25/2012

Now I have the whole manual. Both of these are about 10MB:

TM 11-900
TM 11-900 rotated for readability

My manual is copyright by me, Kris Kirby. All rights reserved. This manual may be used for non-commercial purposes only. This manual may not be included in derivative works, or compilations of works (i.e.: CDROM collections), or sold on eBay either in print form or on electronic media.

Please contact me if you have a desire to include the manual in distributed works; we may be able to come to an understanding.

I have much higher quality versions of this document already scanned at over 150 DPI, uncompressed. I'd be happy to work out a license for the higher quality works and/or commercial uses.

DRM hasn't been included in these works because I believe strongly in portability. My original manual was printed/produced in the 1940s. It has survived this long, but it will not survive forever. It is my desire to preserve this information, but not exploit it. It took me a long time to scan, process, and so on. And you'll note from the document that I didn't always get everything straight. I don't seek to recoup my time, energy, and hard drive space. But there is a significant personal investment of each of those.

If there's something you can't read on a page, let me know. I may be able to re-scan the page and change the documents. It was stored in the accessory box of the generator, so it is filthy and does tend to leave specks behind on the scanner.

I can be reached at kris ]a()t[ catonic d0t us.

Damn Cool Antennas

http://www.athena-gatech.org/research/3DINTEGRATION-MATERIALS/index.html

Neat stuff.

Ramsey FTR-146

I've just received an email from Ed over a Ramsey in the technical support area. I waited a few days and he patiently scanned the last manual they had for the FTR-146!

I'm sure they'll get a copy of it up on the website soon, but I'll mirror it here as well.
 
FTR-146.pdf

So the basic routine for programming is:

Take the Desired Frequency
Subtract 143.000 MHz from it
Multiply by 100.

For example:


146.940 MHz
-143.000 MHz
-------------------
    3.940
    x 100
---------------
       394


Then you set the PLL divider / diodes to 394 in binary.

394
-256
-----
138
-128
------
10
-8
-----
2

So you put diodes into 256, 128, 8, and 2. (and -600KHz if you're using the repeater).

That's it.

Packet Radio / TNCs

Thoughts/observations:

Speed is important. Baud rates are limited by law, but Baud doesn't equal signal rate; baud is the baseband. Using QPSK, you can double throughput, and you can still throw bits away in FEC if you need to. Phil Karn is a huge proponent of this and for good reason; he had a critical role in developing Qualcomm's satellite-based terminal systems used by truckers everywhere.

6m might be good for this, but easily obtained radios (Motorola Syntor Xs, GE Deltas, etc.) are starting to disappear. Also, they are large, and without a small TNC/node hardware that fits inside the radio, there is little reason to deploy equipment because more parts can break. Of course, the radios themselves need about 30A on transmit, so that would also need to be dealt with in a manner that doesn't impact size and site power requirements.

There's very little reason why we can't push soundcard packet into smaller systems like the Alix line of micro-PCs. We can dedicate an Arduino to software packet detection, another to node/routing, and pass the data on to a host if need be. Or we can load the sound-card engine into memory as a TSR and boot the thing using DOS and a G8BPQ stack. With TNC-X, a KISS TNC, it's possible to do that and more. Ideally, speeds upwards of 19200 and full-duplex are desired. However, full-duplex generally requires real hardware on the "node" side of things, and duplexers are a generally fixed commodity.

Also, proxy ARP may be a better way to use TCP/IP over AX.25, stealing information directly from the NETROM maps or something. For instance, my state net is 44.100.x.x, but good luck trying to actually get any of that to route outside of Mirrorshades.ucsd.edu.

Really, something closer to the mesh-networking systems used for the next generation wireless networking systems would be better. To gracefully handle losing a node and multiple routes present in the network stack. You can't really do that on a Z80 with 16k of RAM at 10 MHz.

IPv6 needs to be implemented at some point, with a graceful handling of IPv6 addresses to allow for compacting unnecessary zeros.

Software TNCs/Minimal TNCs:

AVR:
http://www.byonics.com/tinytrak4/
http://vk7hse.hobby-site.org/wiki/index.php/Main_Page
http://www.garydion.com/projects/whereavr/

Arduino:
http://www.adafruit.com/blog/2010/06/11/aprs-radio-shield-for-arduino/
http://www.arduino.cc/cgi-bin/yabb2/YaBB.pl?num=1232745624
http://dangerousprototypes.com/2011/01/31/packet-radio-and-the-arduino-radio-shield/
http://wiki.argentdata.com/index.php?title=Main_Page
http://forums.adafruit.com/viewtopic.php?f=10&t=11763
https://sites.google.com/site/ki4mcw/Home/arduino-tnc
http://mhvlug.org/pipermail/mhvlug/2011-April/031359.html

PIC:
http://www.ringolake.com/pic_proj/pic_index.html

HF:
http://www.brazoriacountyares.org/winlink-collection/AGW/PE%20Pro/pehelp/6hf.htm
http://www.tapr.org/pr_intro.html
http://wa8lmf.net/ham/30m-magloop-ant.htm

Packet general:
http://www.kc2rlm.info/soundcardpacket/6modes.htm
http://www.enide.net/webcms/index.php?page=wb8wga-tnc
http://www.amsat.org/amsat/articles/kd2bd/9k6modem/
http://nonbovine-ruminations.blogspot.com/2008/05/ham-radio-internet-and-cell-phone.html
Buck's articles: http://www.buxcomm.com/catalog/ (look down on the left-hand side)
GMSK:
http://www.southgatearc.org/articles/highspeedpacket.htm

Ham general:
http://hamlib.sourceforge.net/

Nodes:
http://digined.pe1mew.nl/?Introduction
http://www.ir3ip.net/iw3fqg/uidigi-e.htm
 -TheNet:
http://vectorbd.com/bfd/thenet/index.html
http://nl3asd.tripod.com/thenet.html
http://g8kbb.roberts-family-home.co.uk/html/thenet_x-1j.html
http://servv89pn0aj.sn.sourcedns.com/~gbpprorg/Radio_Mods/MISC/PACKET/PACKX09.TXT
http://kf8kk.com/packet/jnos-linux/thenet-ops-1.htm
- IP use in TheNet nodes:
http://www.w7eca.net/forum/viewtopic.php?f=84&t=444
 - TheNet replaced by NOS:
http://62.49.17.234/thenet.htm
 - JNOS:
 INP something. I dunno...
edit: Ah, here it is. A European internode protocol:
http://dl6mpg.net/nordlink/ftp/pub/documentation/INP/inp3.pdf
 http://www.langelaar.net/projects/jnos2/documents/inp2011.txt
http://www.langelaar.net/projects/jnos2/news/
Intro to NOS: (packet sizes, numbers)
http://www.febo.com/hamdocs/intronos.html
http://kf8kk.com/packet/jnos-linux/whetting/whetting.htm
FlexNet:
http://www.afthd.tu-darmstadt.de/~flexnet/

DX cluster:
http://www.ab5k.net/Home.aspx
http://www.dxcluster.org/main/index.html

Antennas:
http://wa8lmf.net/ham/30m-magloop-ant.htm


Old Huntspac stuff:

http://www.qsl.net/n8deu/huntspac.htm

Most of the older stuff I saw fall out of use as people got older and fell into different modes / cliques / clubs



Amateur Linking and Backhaul

I have given a lot of thought to this. One of the issues with linking repeaters in the amateur world is that we're trying to do things that have already been done, years and years ago. Fortunately, all of that wisdom and experience is available at our fingertips, or out of the mouths of those old enough to remember and willing to speak. All the way back to Ma Bell's Long Lines microwave relay system, there have been various concepts introduced.

Long Lines was a channelized system which relied on translation (the block conversion of a group of frequencies to another group of frequencies) of existing channels for repeating or relaying and added a capability called "add/drop." Add/drop is a term still used in the telco world on circuits and certain types of gear. Long Lines was largely based on frequency-division multiplexing (FDM) of 300-4000Hz single-sideband (SSB) amplitude modulation (AM) signals. The resulting SSB signals were several tens or hundreds of kilohertz wide, depending on the number of channels the radio circuit was designed to carry. SSB is also subject to noise, and like most baseband or analog communications designs, requires equalization at multiple locations in the circuit equipment. Cable TV (CATV) experiences this as well. The increased attenuation at higher frequencies is referred to as "tilt". CATV amplifiers are designed with a "tilt" adjustment that allows for the higher frequencies to be amplified more than the lower frequencies to compensate for tilt losses. Later, Long Lines was transitioned in part to Digital Radio, using full-duplex DS3 radios costing $50,000 or more to connect cities or sites along the route.

Add/drop telephony refers to a piece of gear or circuit where channels are picked off and either not replaced, or replaced with other information. Synchronous and Asyncronous circuits alike may be configured for add/drop simply by adding equipment. In the sense of Long Lines, the Add/Drop functionality was not added by equipment; it was a function of that switching office itself. Microwave channels could be received on one dish or horn, translated to a different set of frequencies, and transmitted out another horn in a different direction. Likewise, a copper circuit could terminate at the Long Lines facility and be converted to a microwave channel and "Added" to the network. In the reverse sense, a microwave channel could terminate at the Long Lines Office and be converted (Dropped) to a copper circuit leaving the Central Office (CO). In T-series circuits, individual circuits may be added or dropped out of the larger group present. Modern equipment like the Adtran Atlas can do this on the DS0 (56K or 64K circuit) boundaries from T3s or DS3s. Other equipment, made by manufacturers such as ADC-Kentrox, may add/drop a single or multiple DS0 channels from a T-series (T1, T1c) circuit to allow combining virtual circuits over a physical copper loop from the telephone company.

The use of digital radios by Long Lines and another companies in the long-distance markets was supported by the marketing the systems allowed the company to do. Many systems which switched to PCM coding for voice immediately met with quieter, noise immune circuits. But the common plague of digital electronics still pervaded the space -- if the circuit was up and correctly configured, it would be error free. If the circuit was down or incorrectly configured, it may take months before the actual cause of the problem is located and remedied. Handling audio in the digital domain also freed technicians from the tedious labor associated with individually aligning circuits and amplifiers as well as equalization requirements for each stage of the equipment.

Several manufacturers make full-duplex digital T1 (24-channel) modems which allow telephone circuits to be extended many miles beyond the initial termination point. One well known radio station uses this method to allow the sales, management, and studio staff to have access to a nearby larger LATA (telephone calling zone) without paying long distance. This sort of connectivity, along with the ability to add and drop circuits at will, allows one to selectively "dial up" one or more channels of voice or data to a remote endpoint. In the data world, it is quite easy to inversely multiplex a larger, faster rate signal into multiple slower channel circuits to achieve higher throughput. This has been used since the infancy of the internet and the Apollo Space Program for transmitting data, telemetry, and video across analog circuits.

One of the larger issues and concerns surrounding amateur radio is the use of shared tower space. Tower space is often at a premium, and is very difficult to find for cheap or free. Furthermore, there is a complicated issue with respect to the Part 15 bands, primarily 902-928MHz, 2.4 - 2.45 MHz, and 5.3/5.8GHz. Many of these bands coincide with amateur radio bands, in which amateur radio operators are granted the immunity from other operators in the band who -- being unlicensed -- must tolerate interference from licensed users of the spectrum: amateur radio operators. Where this turns into a conflict of interest is that the Part 15 band user may want access to the tower and may mount commodity video, data or other equipment on the tower which may cause interference to users in that band who are licensed or otherwise permitted use of those frequencies. Moreover, the Part 15 user is a for-profit enterprise and is paying rent for the tower, while the amateur radio operators, who claim a higher priority to the frequencies desired by the Part 15 operator, are not paying for rent. The simplest solution in the eyes of the site manager is to tell the hams to pack it and leave -- money talks. Hams are strictly forbidden from engaging in for-profit enterprising using amateur radio, so as long as that rule stands, ham equipment on towers will cost a large amount of money to setup and maintain. The solution, therefore, is to avoid frequencies which are shared with other users of the site or that might cause interference to other paying users of the site. This basically confines a ham to bands which are solely the domain of radio amateurs, or shared with federal agencies. 222-225MHz (1.25m) is one these bands, as is 420-450MHz (70cm).

The FCC Rules, Part 97 limit the speed and amount of bandwidth that transmissions at or above certain frequencies may occupy. At the time of this writing, the following limits apply for 28MHz (10m) and up:

  • 10m: 1200 baud signaling rate, FSK may not exceed 1KHz
  • 6m: 19.6 kbaud signaling rate, 20KHz emission width
  • 2m: 19.6 kbaud signaling rate, 20KHz emission width
  • 1.25m: 56kbaud signaling rate, 100KHz emission width
  • 70cm: 56kbaud signalling rate, 100KHz emission width
What strikes my eye about this is the rate, specified as "symbol rate", particularly in bauds. A BPSK signal, operating at 9600 baud, transfers data at 9600 bits per second, at one bit per hertz of occupied bandwidth. QPSK on the other hand, transmits two data bits per symbol, doubling the capacity of the same 9600 Hz of occupied frequency. Switching over to QAM at higher densities allows for even more bits to be packed into a given space. Using QAM or QPSK requires linear amplifiers and different modulation methods to produce them. This adds complexity, however it extends the most precious resource we have -- limited bandwidth. By using 4/pi QAM on a 56Kbaud circuit occupying around 100KHz, 224Kbit/s of data may be transmitted using linear amplifiers. At these widths, duplexers may be used to fine-tune filters to keep noise out of the systems, as well as prevent possible interference from or to other stations on the same mountaintop. Additionally, being down in the amateur radio bands prevents a fight over who stays and who pays to use a given section of vertical real estate. Since the amateur radio operator isn't sharing frequencies with the unlicensed or commercial operator, this situation should not exist.

In short, we owe it to ourselves, as amateur radio operators, to explore dense data modulation techniques and apply them to amateur radio. We have had a thirst for bandwidth dating back decades; now we finally have commodity modems that make achieving those bandwidths possible using limited RF. Just look at Digital TV: 19 or 23Mbit/s in 6MHz. Twenty years ago, in 1990, that would have been astounding! And now we have that on ever single person's set top TV.

Improvized Power Loads

Being something of a home-bound hacker without a lab other than my own equipment, I find it's sometimes necessary to improvise equipment using other stuff. After replacing batteries in a UPS, I needed to check everything out for loose connections, unusual sources of heat, and so on. It took a lot of effort to take the UPS apart, and I didn't want to have to take it back apart or have any issues inside that would crop up later.

When testing a UPS, or another other power generating device, a resistor is the best type of load to use. Since it's non-reactive, the impedance of the resistor equals its resistance across almost the entire spectrum.

Here's where we go crazy...

First, I tried the toaster oven. Fail. The toaster oven is 1400W. UPS is nominally about 1500 VA, which works up as a few short, because most computer UPSes are overrated in VA because most PC power supplies don't have a .99 power factor.  Good test of the overload capacity.

The portable heater didn't work out for one reason or another, 750 or 1500W. So I was left searching for something that would do the job. The microwave, at 1500W, was also out of the picture. The electric skillet was an amazing 1200W! Finally, I settled on the one obvious solution for some UPS runtime: the rice cooker / steamer.

The steamer's nameplate said 650W at 120V. This was a perfect load for the UPS, as it didn't exceed 1000VA or 1000W, allowing me some actual run-time with the UPS. Since the steamer works through phase change, the actual "output' of the device in steam wouldn't be very much. There would, however be a few cups of hot water in the bottom.

So remember the next time you need to do a test, what heaters you're surrounded with. Just because a heater is designed for 120V, doesn't mean you can't apply it at a lower voltage.

650W / 120V = 5.4167 A. 120V / 5.4167A = 22.154 ohms.

Likewise, were one to attach an 8-ohm speaker across the 120V line, it would need to dissipate 1,800 watts and would trip the breaker on a 15A circuit eventually.

120V / 8-ohms = 15A, 120V * 15A = 1800W.


If you really still want to buy power resistors, and there's no reason not to, you can find them cheaply at Surplus Sales of Nebraska and Fair Radio Sales. Be aware however, that above audio frequencies, impedance may become a factor as the device may start radiating. Just because you can match a HF transmitter to two steamers in series doesn't mean you should use them as an RF dummy load. Gordon West, WB6NOA famously demonstrated this by making contact with a fellow amateur radio operator across the world using a light-bulb as a dummy load.

Tower Work

Do you have an abject fear of heights? Do you quake looking through a glass or metal floor above twenty feet? If so, you probably shouldn't be doing tower work. But if you are, here are a few other things you should be aware of:

If you need a rope to haul anything other than your tools, you need a ground crew.

Just because an antenna and/or bracket moves smoothly on the face of the tower doesn't mean you can lift it, or take it off the tower by yourself.

Never estimate the amount of time it will take to finish a task working alone. It will take longer, and eventually you'll find some point where you will need more power from the ground than you can exert on your own.

I'm going to update this as I go, so look for more lines to be added over time...

FM+ Spread Spectrum + SDR = The Win

I had the thought the other day that, with high resolution DSPs operating at 150MHz, there should be no reason why we can't have a SDR which spits out exactly one cycle of energy on one frequency, then one cycle on another. The discriminator of a receiver should discern this as FM. The thought I had in this, is to do not just this, but to time division multiplex (TDM) the radio itself, and occupy two frequencies at once, several kilohertz apart. As the upper part of the voice signal is 4,000 Hz, there is no reason in my mind why two or more signals can't be transmitted one pulse at a time by switching single pulse frequency around the two target frequencies at rates fast enough to permit them to be interleaved. 8,000 pulses per second is short in a train of pulses that occurs 150,000,000 times per second.

SDSL Pipelines

Or what I know about them now that you don't.

Years and years ago, I got introduced to the Pipeline series of routers by a friend and employer. As a way to save money, he had obtained an Ascend Pipeline P130 and used it to replace a Cisco 3640, saving some $500 a month in router rental. The catch was that I had to figure out how to make it talk to the provider's circuit. This was easier than realized, once I had them on the phone and they told me they could switch the DS1 over to PPP instead of Cisco HDLC. When they did, the circuit came up. A few months later I attempted to the same trick with a new provider network and a different version Lucent Pipeline P130, and had absolutely no success for several days. Finally, I noticed the "Nailed Group" number on the new router was set to "1" and the old, working router was set to "3". I changed the Group over to "3" and got lock on the WAN light.

After much consternation with discovery of more sub-variants of Pipeline Routers than there are species of cacti, I finally succeeded in obtaining two SDSL Pipelines which were close enough in feature set so as to support communication with each other. These routers turned out to be Lucent Cellpipes, specifically the DSL-CELL-50S.

Of course, they proved again to be a source of MUCH consternation, as no setting seemed to convince them to talk to each other. Having worked in a xDSL test lab some years ago for a major chipset manufacturer, I knew these could talk to each other, as I had seen one used as a CO device, and read the specs myself and knew that any device that could provide the head-end side of the circuit. Knowing that ATM was intimately involved in the lower DSL levels, I opted to configure the modems in ATM-VC mode, and configured them to bridge.

Bridging on the Pipeline is a very dodgy proposition. Most of the providers used bridging, which lead to the Pipeline getting a bad reputation. At only 25MHz or so of CPU, the processor would attempt to bridge the entire traffic of the 10Mbit ethernet segment onto the WAN circuit, while copying the incoming WAN circuit data out the ethernet port. The end result was that the CPU was constantly overwhelming, servicing interrupts continuously and trying to IO itself to death. On the other hand, if you forced it to do only IP routing, the little bastards flew, happily updating the interactive terminal interface while responding to SNMP. I once schemed to build a BGP router out of a FreeBSD PC using a pair of Pipeline P130s bridging ethernet to DS1s and terminating the PPP session on the PC using PPPoE. Unfortunately, our network never got large enough for that. Such were the days of the DotComs and the fickle ways of the investor.

Back to configuration of the Pipelines,  I set the DSL layer to communicate using ATM VCs over VPI 8 and VCI 35. In the lab we expressed this as "8.35", or "0.35" depending on what port we were using. On the outside, the inner workings of ATM were too complicated for most people to understand, so the VPI.VCI just became another meaningless setting that HAD to be input exactly for the system to work. After the ATM was configured, I rolled back a step and set the modems to communicate at a single DSL speed ("mode=singlerate") of around 2.3Mbit/s. Now the modems started attempting to lock to each other, but kept dropping the call. I set one to COE, and the other to CPE, then configured the COE unit to only answer the call, and the CPE unit to only call, never to answer. Now the calls were being initiated, but data was not flowing. "What the hell could the problem be?" I asked myself.

Fortunately, the several modems I had gather had documentation. One of them actually had the xDSL specific menu addendum, but all of them explained in a physical layer independent method how Ascend Bridging worked, and how the PPP system was used. "Knowing" how PPP worked and that the circuit was largely free of eavesdropping, I configured the units to use PPP over ATM (PPPoATM) to pass traffic to each other. The PPP system was setup to use PAP, and an arbitrary password was selected and configured into both modems as the recieved (expected) and the sent password. Likewise, I named both of the DSL modem's names (or names under Connections) to "DSLPipe". This way, both modems would ask each other for the same bits of information, allowing either modem to assume the role of COE or CPE depending on how I got them configured.

At this point, I have two modems configured to only use 2.3Mbit/s as a rate, ATM-VC 8.35, PPPoATM, and Ascend Bridging over the PPPoATM. As soon as the call went up, traffic started flowing! Now to start tinkering...

First change was to turn of VJ compression and header compression. This brought throughput up to 2.0Mbit/s instead of 1.024Mbit/s. I left bridging on, as I didn't want to explore networks at this time, since most of my home network was a single LAN anyway, and I lacked other devices I could plug in and test connectivity with. Finally, I set both modems to autobaud, with the baserate set to 2.3Mbit/s and the SDSL data rate set to 2.3Mbit/s. They auto-negotiated as expected, and dealt with the interruptions as expected. Finally, I set the circuit type to Nailed on both sides, so that if either end went down, an attempt would be made to restart the connection.

And the data kept flowing. Not entirely bad for only about four hours of work. And the two modems were connected over an eight foot piece of RJ-11 patch cable.

So now you know what it takes to make those bloody modems work. Next project is to attempt the same with the V.35 Pipelines, which are basically a retarded version of the modem I already described.

Originally, I got these modems to attempt a bidirectional data link over 900MHz. By cutting away the resistive Wheatstone bridge-hybrid in the front end of the modem, it is possible to separate out the receive side from the transmit side. By piping these signals to separate block frequency converters (heterodyning transverters), it is possible to put them on the 900MHz amateur band. The benefit to using DSL modems for this purpose is that they already possess adaptive electronics in programming, which allows them to redistribute the bits as the channel capacity allows. If, for instance, a carrier appears and stays in a given place, the modems may alter speeds or re-profile the channel. This allows me to focus on getting the bits where I want them to go, and fiddling with RF, leaving the DATA layer delegated to prior art.  Thank you for reading what may be my longest post ever.

Also, this information is copyright 2010 by Kris Kirby, and all rights are reserved. You may not use any of this information in support of an eBay auction.


It is important to remember that ATM is involved here, so there is a percentage of that which cannot

MASERs and Why We Don't See Them

Over the past few months, I've given thought as to why we don't see large scale development of MASERs. MASERs, which are like LASERs but generate radio frequency (RF) waves instead of light waves. Both light waves and radio frequency waves are electromagnetic waves. We have high power, coherent lasers used for cutting steel. However, we do not at the present time have high-power MASERs. Initially, I assumed this was because it would be the topic of strictly militaristic endeavors. And while that is true, the body of evidence shows that MASERs didn't develop into high-power variants because of different reasons -- there was no need. Beyond generating a stable frequency reference through cesium or hydrogen emission, there was no need seen in generating a high-power single frequency radio emission. I believe this may be due to the fact that radio waves are seldom coherent, and therefore may have been difficult to excite in a precisely controlled manner. So my conclusion is that the hydrogen and cesium MASERs do exactly what their name implies. The technology is mature. Unfortunately, the lack of a high-power derivative limits the offensive use of the technology, as well as use in communications. And we've gone on to use MASER-like technology to excite hydrogen atoms with magnetic fields -- we call them "MRI", or Magnetic Resonance Imaging.

Understanding I and Q modulation

This article is borne out of an attempt to explain modern modulation methods to a friend who has only heard of AM and FM.

http://education.tm.agilent.com/index.cgi?CONTENT_ID=4

This article explains and illustrates how a difference between the "I"  and "Q" produces modulation. I can be thought of as In phase. Q is short for Quadrature phase, which is the phase difference when measured from a fixed distance of -90 degrees from the I signal.

Is there is fixed shift in frequency, the resulting modulation is a circle. When viewed along the time axis, the circle becomes a spiral. If the spiral moves to the left (left-hand circular polarization), the frequency is higher than the carrier frequency (I leads Q). The converse is also true; the spiral turns to the right if the frequency is lower than the carrier frequency(Q leads I). Either of these signals, viewed on an oscilloscope, appears as a circle turning one direction or the other depending on the leading component.

If you have the ability to modulate the I and Q phases directly, any known modulation may be synthesized using a mathematical expression. AM is perhaps the most simplistic to represent since resulting occupied frequency is symmetrical. Holding the I phase steady in amplitude and phase while varying the frequency of the Q phase results in frequency modulation. It is almost as if varying the I phase amplitude modulates the signal and the Q phase frequency modulates the signal and the two together make any modulation, but this not accurate since QPSK is a phase modulation. It would be more appropriate to say that phase modulating both axes produces QPSK, and through that ability QAM, FM, and AM may be created. For this example I do not distinguish between AFSK, BPSK or FM, nor MSK or GMSK since they are all frequency-modulated modulations despite individual complexities.

One point of note that I have not seen in the books yet is whether or not the QPSK or PSK signals are subject to pre- and de-emphasis of the signal as a result of the modulation. In two-way radio, phase-modulation came about as a cheap way to frequency modulate transmitters, however it has a built-in pre-emphasis of the transmit signal, amplifying higher frequencies more than lower frequencies as a result of the modulation process. In the data world, this causes the resulting signal to occupy more frequency bandwidth than the symbol rates would otherwise imply.  

Here is another good explanation.

Twist-Shift Keying or Primative PSK?

Published, 2009-02-08 @ 21:08:58:

http://maxmcarter.com/twistmod/index.html or Twist-Shift Keying, documents an idea to modulate RF around the axis of the antenna ('boresight') instead of modulating the phase, amplitude, or frequency of the antenna.

I am at present formulating a response to this.

My current feelings are that this is a slow, imprecise version of phase shift keying, wherein the signal is phased at +90 or -90 degrees. This is a confusing idea to consider. According to documentation provided by the author, the transmitted signal is either left-hand circularly polarized, or right-hand circularly polarized. From the perspective of the discriminator, this should mean that the signal is approaching or receding, however one also has to think about the fact or concept that the antenna itself may force the signal to simply not appear at all.

I think that for the purposes of considering this signal or modulation method, it may be easier for the reader to work within the confines of linear polarization. Linear polarization is typically defined as either vertical or horizontal polarization, however one may also implement 45-degree and 135-degree polarization by implementing a dipole rotated to that particular alignment. Again, in simplifying mental math, vertical and horizontal polarization need only be considered because of the theoretical infinite loss caused by a linear polarization mismatch.

Moving further along the above idea, one may interpret that when a signal appears on the vertical plane, said signal is absent from the horizontal plane. This, in effect, makes the polarization modulation a crude form of a binary modulation, not unlike a form of Orthogonal frequency division multiplexing.  In the theoretical world, it is possible to determine four states from the two polarizations and the state of the transmitter for each polarization.  Realistically, this may approach the impossible as polarization may be affected by antenna location and/or nearby reflectors. In a long-distance implementation terrain, atmospheric, and ionospheric effects may render the signal completely unusable without some form of a transmitted reference or a guide signal or encoding. It is the opinion of this author that switching to circular polarization would not rid the signal of polarization modification.


The nice part of this supposed modulation method is that the receiver antenna orientation does not affect the capabilities of the receiver so long as the antenna area of sensitivity is pointed at the transmitter. The receiver is only required to differentiate between the two polarities of circular polarization.

Possible? Yes.
Implementable? Yes.
Practical? No.
A use for two Syntor X transcievers? Yes, better than holding down the dumpster. =)

Update: Now that I've had to opportunity to learn a bit more, I realize why this approach is novel but also useless. Previously, we have not modulated a signal based on polarity because polarity may be altered by the physical terrain the signal travels over and through. On HF, Faraday rotation may occur which causes a signal transmitted with a vertical polarization to be reflected in a different linear polarization, such as one at an arbitrary angle anywhere from zero to ninety degrees. For example, vertical polarization that has been shifted ninety degrees is horizontal polarization. Any shift from 90 to 180 degrees is indistinguishable from a shift of zero to ninety degrees without additional polarization information.  This shift may be resolved using interferometry and a second receiving antenna a fixed distance from the first.

The biggest practical reason I can see for not using Quadrature Polarity Modulation (QPM) is that it is often necessary to cross-polarize an antenna to minimize the effects of an interfering signal. Furthermore, using horizontal polarization for long-distance linking allows the reflections from the terrain to cancel each other out, resulting in less multipath interference at the receiving antenna.

I am certain the QPM was discovered at some point in the infancy of radio. I believe there are a number of factors which contributed to abandonment of this form of modulation.

2013-07-25: Edit and update

The fiber optic industry has, in fact, already done this:

http://blog.catonic.us/kirby/2011%20IEEE%20CTW_Peter%20J.%20Winzer_keynote%20slides.pdf

and did QPSK over it.

Another Use for Stella

Another use for STELLA would be to improve the intermodulation characteristics of an amplifier. Any time you transmit two carriers into a satellite, it is possible to have the satellite's transponser -- or any linear amplifier -- generate a large number or mixing products. Using a system such as STELLA, an error signal may be generated. This error signal would contain all of the carriers that are desired to be notched out of the signal, out of phase with the signal by 180-degrees, so that when the two signals are combined (or in the digital domain, complex multiplied), the resulting signal will be devoid of the extraneous carriers. A technique not unlike this is used in modern TV (OFDM) transmitters to restore linearity to the signal as the amplifier moves through non-linear points in the performance curve of the amplifier.

Amateur Radio's Political Side

Kris: so, are the hams there really just antisocial or what?
Tyler:
not really sure
Tyler:
they seem all nice over the air, but in person they are stand offish
Kris:
most radio folks are nerds or geeks, or just idiots
Kris:
or ZOMG ARES whackers
Tyler:
heh
Kris:
you know what I mean about ZOMG ARES whackers, right?
Tyler:
you have shown me photos
Kris:
some of them don't have the distinctive dress

There are some people who got into amateur radio for one crazy reason or another
. I got in because it was neat at the time, and I've met some neat people. By neat, I mean real engineers that I can bounce ideas off of and keep the crazy from infecting me or becoming part of another plan, only to have it unravel later due to a missed detail  Other people got in because of a want or need to spot weather or report weather to someone or something who cared.  Still others got into it because of a misguided desire to serve the public  It's the last category that causes the most trouble, and also provides the most visibility. Terms like "served agencies" further muddle the waters, and the attitudes of these people cause immense amounts of ill will between individuals and groups.

Consider, for example, a group that comes into or forms in a given area for the purpose of "emergency communication." The group has no repeater, no group resources to speak of, and only a marginal source of organization -- a monthly or weekly meeting that not every member attends. The group surveys the local airwaves and determines that there is a repeater that serves their needs. The group declares that they will meet on that repeater frequency for the purposes of furthering that function -- emergency communication -- and drills thereof. This strikes the owner of the repeater as presumptuous, because, like many other repeater owners, he has invested much time, energy and money in establishing that repeater, and the relationship with the repeater's site owner to make that repeater a possibility. This emergency communications group has not taken part in any of the above action until it involved keying a microphone on a radio that a group member himself owned.

Sadly, my example is actually generic; this situation has repeated itself numerous times throughout the country. When I initially wrote this document, another group was in mind. I can now say however, as a repeater trustee, that this situation has happened to myself. Some of the supposed emergency communications groups causing real harm and discomfort between the repeater owner and the site owner. These site owners are often not amateur radio operators, and don't understand the finer details of FCC Part 97, or what amateur radio is about. As a result, these persons, in a state of confusion, may take the easiest course of action in the face of controversy -- to order the repeater owner to come and get his equipment and vacate the premises effective immediately.  

There has been a movement in ham radio as if to find a purpose for spending all this money to either justify buying newer toys, or to justify taking tax money from federal agencies. Public safety has already proven to be a dangerous bedfellow; one need only look at the ReconRobotics case to realize why. I was rather amazed to see at one hamfest where even some of the most ardent of the "served agencies" crowd quickly about-faced and said "let them get their own frequencies."  With great freedom, comes great responsibilities. We asked for spectrum. In exchange for large swaths of space, we are responsible for self-policing and self-regulation, coordination of activities with other operators to avoid mutual interference, and generally interface with the FCC on a lesser basis than any of the other licensees in any of the other services that the FCC regulates.  We must be careful of who we choose to associate with, and avoid those who would harm our hobby. Few hams have the resources to stand as Free Men, and challenge bodies of government when missteps are taken or about to be taken. For this reason, we must continue to be as vocal as possible to keep our issues alive before the FCC, and to assure the FCC that we are doing everything in our power to both protect and use the spectrum afforded to us.

We must also strive to self-educate and redirect those hams who are getting too fanatical. It is certainly one thing to be nuts over contests, EME, or microwave. But to have this need to define one's purpose over or through an interaction with a group whose only interest in radio is that they carry one on their side for communication is just preposterous.

A Remote Control Station

My extensive research has pretty well proven that there's no single source -- no product, software or hardware, that is manufactured and carried by one of the large ham radio stores that may be purchased through a purchase order.

This is a niche, and there is a void.

I+Q MoDem

http://www.g6lvb.com/Articles/STELLA/index.htm
Documents one British ham's implementation of an RF equalizer network using a DSP and a pair of DAC/ADC stages at bandwidths of about 96KHz. What's interesting about this is several points. The implementation allows for spreading power evenly throughout a transponder, allowing the transponder to remain linear in the face of a signal that would otherwise swamp, dominate and further modulate the output. This could also be used to allow the transponder linearity to be matched to the current loading, keeping the transponder linear even in the face of extremely high channel loading. This sort of situation would be seen in the event of transponder oversell in the commercial world, or through the use of spread-spectrum and/or wide bandwidth signals such as QPSK, QAM, or OFDM.

What's further interesting about this system is that it allows for "audio bandwidth" processing of signals. Virtually every PC has a sound card that may capture a sixteen-bit signal at up to 44.1KHz, some as fast as 48KHz, and others at 96KHz with a depth of 24 bits. As far as audio performance is concerned, the bit levels are directly convertable to RF performance. Both are a means to describe the dynamic capabilities of the capture and reproduction aspects of the card. In the audio world, There is typically an upper limit of somewhere around +12 to +24dB, and the noise floor somewhere at -110dB to -130dB, with a reference of 0dB or +4dB for a professional sound card (96KHz, 24 bit). What these numbers translate into as voltage levels largely depends on how the manufacturer designed the card and what form of termination that manufacturer had in mind. A typical impedance would be 600 ohms, but some cards have a drive impedance of 100 ohms. 100 ohms is not a terrible match to 50 ohms, which is what may be expected to drive mixers and other RF gear at the baseband level. To assure there are no issues at the baseband level, I would recommend making an effort to match the impedance of each device or at least provide a buffer or attenuator to lessen the impact of an impedance mismatch.

The December 2008 issue of QST http://www.arrl.org/qst/?month=12&year=2008#toc contains an article entitled: "A Modular Reciever for Exploring The LF/VLF Bands [part 2]", which details and implementation of a Tayloe Detector. A Tayloe detector is an interesting piece of RF work in and of itself since it's not a detector (or rectifier) in the conventional sense, but a commutating detector. Now there's an term you don't hear bantered about often in RF or EE circles unless you're designing large civil projects like power generation or electric motor based facilities.  In essence, four switches, each one of which matches to the four phases (0, -90, -180, and -270) are switched to one of two outputs. This provides a means for detecting the sine and cosine components of the signal which are otherwise known as I and Q. The switches themselves function as the rectifier would, except with much narrower windows for conduction. This circuit would be almost impossible to implement conventionally without SCRs and a ninety-degree phase delay line. Here's a link on the Tayloe detector: http://rfdesign.com/mag/radio_lownoise_highperformance_zero/  The Tayloe detector is just another method for demodulating the RF once it's been downconverted to an IF frequency to pull out I and Q signals. From there, you step into the digital domain for further processing, or simply right back out to make a transponder.

If you capture the signals using a plain-old PC sound card, it's possible to capture a signal of 22KHz, because of the Nyqist limit. However, using a PC-sound card limits the dynamic range -- or how large a signal the signal processor can handle, or how soft a signal it can receive (reciever sensitivity).

Capturing I and Q from the baseband level allows any signal to be repeated, not just FM, AM, SSB, or D-Star for example.
 

TDM, The only way to go

I have read to so much about various VoIP systems being pressed into radio use, but many of them ignore time as a requirement for audio. The chief reason why time is so important is that in a voted or simulcast system, you must have constant time figures across the link, or the noise comparators will be comparing dissimilar signals, resulting in unpredictable voting behavior. It may be simple enough to build a conventional FM repeater to pass a signal between two points, however the repeater has the notable disadvantage of "coloring" the audio passed through it. The audio that is recieved is subjected to the reciever's de-emphasis network as well as the transmitter's pre-emphasis (which is not a network, but a side-effect of phase- or frequency-modulation). This is documented and explained in depth here: http://www.repeater-builder.com/tech-info/flataudio.html However, due to this constant cycle of audio processing and deprocessing, the passed audio is "colored" after crossing each stage. This requires equalization of the path, to restore the audio back to something resembling a "flat" path, which does not express such characteristics. The simple way to avoid all of this complexity is to digitize the audio as soon as possible, and send it over the air as a digital signal. This is what Ma Bell started doing in the 1970s, and continues to do to this day.

Furthermore, if E&M signalling is used in the link, with E&M four-wire line cards, there is a seperate transmit, reciever and signalling pair presented to the end user. E&M signalling is bi-directional, so there is a local switch input, and a remote relay output. These two signals are very simple to connect to a reciever and transmitter, allowing carrier squelch detect to be sent and push-to-talk (PTT) signals to be sent!

There is Part 15 Microwave equipment manufactured which support a full-duplex T1 (1.544Mbit/s) or E1 (2.048Mbit/s) radio link. The equipment was manufactured by Western Multiplex, and put in use all over the country. One well known radio station located in Athens, AL used a T1 radio link to get a Huntsville-local phone number, while having a physical station location that is two LATAs away. Even if the station had subscribed to the "Area Calling" service from the telephone company to defray the cost of what would have otherwise been long-distance telephone calls to Huntsville, the stations's listeners would have had to dial a phone number that was not in the local Huntsville calling area to reach the station. Using the microwave TDM link prevented this, and subjected the station to telephony outages when the link failed during the projected downtime for the wireless link. The station management felt at that time (and still to this day) that the outages were an acceptable trade-off.

 
In the modern telecommunications arena, the sort of circuit is frequently implemented using wireless OC-3 hardware carrying DS1 trunks, as the ATM circuit may be seamlessly and automatically rerouted if a physical link goes down. One local public service provider has implemented such equipment in the fashion of a large ring around a tri-county area, and is using it to provide radio backhaul to a centralized switching location as well as supporting remote bases in that tri-county area. Since the system is full-duplex, the audio delays are static in length, and audio circuit response is flat. Predictably, the system is a smash hit. To date, the only complaints I've heard from the operators are that the combiner networks used to combine ten transmitters into a single antenna have too much loss. Unfortunately, big cavity resonators have quite a bit of loss outside of thier resonant frequency.

In amateur radio, the attempts to implement a similar solution have entirely too many cut corners, brought about largely by the availablity of affordable equipment, and a lack of solutions to fill a smaller need. For instance, in planning to accomplish a goal of a radio link between Birmingham, AL and Huntsville, AL, it is necessary to make at least one hop in the middle of the path, or two. Ma Bell chose to make five, but she was picking up other communities along the way. I have identified a site that would be perfect as a midpoint, however, I do not need twenty-four channels of voice, nor do I think that I could "resell" any of those channels to other amateur radio operators, or the owner of the tower on which the equipment would have to sit. Furthermore, the need to have two different frequencies complicates matters, as each T1 radio would need to have a solid dish, and hopefully nothing in near sight of the dish that would allow the signal to be bounced back into the other dish. The typical ham implementation uses 802.11b, 802.11g, or 802.11a for a wireless backhaul, each of which is half-duplex! This results in unpredictable delays, not to mention much wasted "air bandwidth" waiting for the reciever to lock once the transmitter is on the air. And since all transciever control is handled in the protocol and driver levels, it's virtually impossible for the end-user to tune the transmit-to-recieve times the way that a packet radio (AX.25) user may by altering PACLEN and TXDelay. 

A perfect solution for this problem would be a radio link that had an ISDN or double ISDN width over the air. This would permit call-status information to be sent, as well as two full-duplex audio paths. If one so desired, it would be possible to use the entire bandwidth to support a data transmission at 128 to 144 kbit/s. This would obviously require about 256KHz worth of radio bandwidth, not including guardbands for an MSK implementation. While QPSK and QAM may be used to lessen the radio spectrum requirements, delay is introduced as well as a requirement to use linear amplifiers. Using a linear amplifier increases cost, reduces sources of the amplifier (if pulled from other surplus or used equipment) and increases operational costs, as the amplifier efficency drops from eighty percent to thirty to fifty percent. GMSK is a possible modulation solution which may be implemented in such a fashion as to allow class-C amplifiers to be used.

GMSK achieves approximately 0.7 bits per Hz of occupied bandwidth, which for a 144kbit/s data channel (2B + 1D, ISDN standard) results in about 206KHz. Push that out to about 225KHz to allow a little guard banding, then realize you've got to have the same coming back in on another channel. Provided that the baseband doesn't experience a large group delays, and you can get filters wide enough, it should be possible to retrofit a two-way radio to handle the RF duties. From a signalling perspective, ISDN is clunky at best. There's no less than 500 different parameters, and you've got to order the standard from Bellcore or ITU in order to choose which ones to ignore and which ones to use. It would be a far better idea to pick two channels out of the twenty-four in a DS1 signal and send those over the link, rather than attempt to implement ISDN. Personally, I doubt that an implementer would suceed in finding E&M modules for an ISDN station adapter. It would be simple to use a DS1 channel bank or CSU/DSU with an E&M module or an Add/Drop CSU/DSU. The difficulty here is in implementing a solution cheaply and efficently, given the bar of providing a DS1 interface (and the 1.5Mbit/s signalling) or finding DS1 channel banks for cheap or free. 

D-Star Repeaters on the cheap

Few amateur radio operators who are using D-Star have bothered to read the specifications of the protocol, or understand the mechanism which they are using. D-Star, over the air, is merely 4800 bps GMSK data. To sucessfully repeat D-Star, one does not need a complicated controller, or even a codec chip from DVSI! All you need is to change the callsigns on the fly in the protocol field, and copy the data from the "IN" modem to the "OUT" modem. You want a courtesy tone? Well, send a beep over the air on your HT, and record the data. When the data is played back over the air, you now have a courtesy tone!  And of course, you may use the same method to build a "dit" and "dah", as well as an unmodulated "tone" and make a MCW identifier.

Look in this space for some links to projects similar to this.

Update: Promised links:

http://www.qsl.net/kb9mwr/projects/voip/plan.html VoIP and Ham Radio
http://d-star.dyndns.org/rig.html.en D-Star modem, will interface to any packet rig.
http://d-star.dyndns.org/node_adapter.html.en D-Star modem capable of repeat.
http://www.moetronix.com/dstar/ Home-made D-Star transceiver implementation.
http://opendstar.org/ Under construction.

Update: The D-Star Hotspot now exists. This is a 10mW device which puts out a signal at two meter frequencies and receives on the same and converts to packet data for direct entry into the D-Star data backbone network.




Discone antennas

All antennas are compromises. While the discone antenna promises on paper to be an extremely broadband antenna, manufacturer's claims of capability do not match the common man's understanding of assumed radiation patterns for this antenna. It is necessary study the antenna patterns carefully to understand conventional shortcomings and depending on desired range, to modify the antenna to achieve true broad bandwidth.

When using a discone antenna, few people are aware that the pattern of the antenna diverges from the accepted pattern of a dipole over the broad bandwidth of the antenna. It is possible, through the use of traps, to limit the operation of the antenna to one or more ranges of frequency, thus permitting the antenna to perform in a primary mode instead of a secondary mode.

The secondary nodes are marked by a pronounced null overhead, impacting the vertical pattern and reducing the effectiveness of the antenna as a 0-90 degree aperture. This is caused by the antenna's length approaching a full wavelength across the top of the antenna, a far from the idealized dipole nature of the disc, which is designed so as to be one-quarter wavelength in radius and one-half wavelength in diameter.

The tertiary nodes are marked by nulls in the elevation pattern along the horizontal axis.  These are caused by the antenna length exceeding 180-degrees in electrical wavelength, causing nulls in the pattern to be reflected off of the conical counterpoise. The reflections of the nulls in combination with the nulls cause the pattern to distort sufficiently so as to create nulls in the normally primary axis of radiation -- the horizontal axis.

Here is a computer model of a discone which is designed for 50 - 200MHz. I believe this antenna model ships with copies of MMANA-GAL.
discone1.PNG 
This is a simple discone, which does appear to have a gap between the disc and the cone. This gap, and the wire associated with it, will throw off some of the modeling by causing a slight directionality.

Modeled for 70 MHz, one can see the lop-sided pattern this antenna has. The red line is the vertical field generated around the vertical axis; the blue line is the horizontally polarized field. 
discone-70mhz-pattern.PNG
When the calculations are shifted up in frequency, one can see the pattern shifting around:

145MHz:
discone-145mhz-pattern.PNG
200 MHz:
discone-200mhz-pattern.PNG
400 MHz:
discone-400mhz-pattern.PNG
800 MHz:
discone-800mhz-pattern.PNG

Clearly, as the frequency increases, the pattern changes greatly. However, this antenna has a strange pattern to begin with, so here's a model I created that is smaller, and simpler:

discone-cut.PNG

The disc is one-half wavelength in diameter for approximately 145MHz. In effect, this antenna is somewhere between a discone and a biconical antenna. Since this antenna exists in only one axis, there is a deficiency in the other horizontal axis. Again, the vertical pattern is the red line (about the vertical axis), the horizontal pattern is the blue line.

Here's the pattern, resonant at 145MHz:
discone-cut-resonant-145.PNG
50MHz:
discone-cut-resonant-50.PNG
70MHz:
discone-cut-resonant-70.PNG
145MHz:
discone-cut-resonant-145.PNG
450MHz:
discone-cut-resonant-450.PNGBecause there is sufficient overlap between the horizontal and vertical patterns, here is the total resulting pattern:
discone-cut-resonant-450-total.PNG
900MHz:
discone-cut-resonant-900.PNGAgain, the total RF pattern:
discone-cut-resonant-900-total.PNGAs plainly shown in the models, the pattern starts twisting and changing as the frequency increases. This is because the antenna changes patterns from a half-wave dipole (90 degrees) to a five-eight-ths wavelength dipole, to a 180-degree or longer length antenna. Antennas longer than 180-degrees electrical length (total length: one wavelength and over) run into traveling wave theory, which causes the pattern the cancel unless sections are separated and phased together with respect to specific frequencies. Note however that I did not model a discone on both the X and Y axes. This further explains why the above patterns are odd looking. A true discone would have additional lobes matching those along the X-axis on the Y-axis. 

Putting traps on the diagonal wires as well as the top radials will allow the discone to remain in the resonant mode of radiation (half-wavelength) for a given frequency. However, the presence of traps in the antenna will increase losses at lower frequencies, as the lowest frequency has to traverse all traps to be radiated. Conversely, the higher frequencies will be trapped within the drive point and the trap of interest, where coax losses will determine the power radiated along with the radiation resistance.




Understanding antennas

A log periodic and a helical antenna have so much in common, it in positively unfunny. For instance, if you take and look at a log periodic and a helical, you see they don't really seem to have much in common. If you build a pair of crossed log periodics, you'll start to see the resemblance: The crossed log periodics is merely missing the spiral wire of the helix. The lack of presence of this wire allows the crossed log periodics to be used in virtually any mode, provided proper phasing is performed.

Phasing these two antennas is the largest challenge in implementing a broadband, multifrequency solution. To cover more than one octave requires that the phasing network function across that octave. Most phasing networks are made or designed from Wilkenson dividers. A Wilkenson divider relies on quarter-wavelength sections of coax at an impedance which is typically above that of the input impedance. Because the divider is a resonant structure, it is a frequency dependent device. One makes a non-frequency sensitive divider using a hybrid combiner.

D-Star Multiplexing

A D-Star radio signal is approximately 6.25 KHz wide, much narrower than the occupied bandwidth of a -/+4.5KHz FM signal (which is somewhere in the neighborhood of 9KHz). It should be possible to, within a 20KHz channel spacing, allocate two D-Star repeaters by offsetting either repeater from the center frequecy by 25 or 50KHz.

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