[digitalradio] Re: New to Digital HF -- PACTOR setup and hardware maybe needed???

2006-08-28 Thread Dave Bernstein
The issue is control over the operating system's scheduling 
decisions. There are real-time versions of Linux that are comparable 
in this dimension to the firmware running in a TNC; given sufficient 
CPU horsepower, a Pactor-2 or Pactor-3 implementation on realtime 
Linux is feasible.

The Windows scheduler cannot be adequately controlled for use timing-
critical applications; as a result, the response time requirements of 
protocols like Pactor-2 and Pactor-3 cannot be guaranteed, no matter 
how fast the CPU. Generating CW with consistent timing is a challenge 
on this platform.

73,

Dave, AA6YQ

--- In digitalradio@yahoogroups.com, John Becker [EMAIL PROTECTED] wrote:

 It was on a linux system
 But that does not matter.
 The problem is EVERY time the computer thinks
 what do I need to do now - the timing is lost and so
 is the link.
 
 At 09:53 AM 8/27/2006, you wrote:
 
 
 Have you tried it with an up-scale sound card on a
 computer equipped with both a fast processor and a
 lot of RAM?
 
 What was the interface between PC and rig?  USB 1.0,
 1.1, 2.0, Serial, Parallel, other?
 
 I am wondering if the rapidly improving PC hardware
 may have solved the speed problem?
 
 Also, which OS did you test, please?  Apple, the
 MS version of windows (XP?), Linux?
 
 I am not questioning your observations just am curious
 as to the testing context.
 
 --
 
 Thanks!  73,
 doc, KD4E
 ... somewhere in FL
 URL:  bibleseven (dot) com
 
 
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[digitalradio] Re: ALE QRM is minimal

2006-08-28 Thread Dave Bernstein
The documentation in http://hflink.com/ recommends that a station 
transmit a 20-30 second sounding hourly on each frequency. 

Below, Bonnie says In amateur radio ALE, there is only one pilot 
channel per ham band where repetitive sounding (station ID) happens 
on a regular basis.

How many stations can be sounding a band's pilot channel before it 
saturates? Lets assume the best case, which is that each stations 
sounds for 20 seconds each hour. If the channel is perfectly 
utilized, it can handle (60*60)/20 = 180 independent stations 

180 non-synchronized stations attempting to sound one frequency for 
20 seconds each hour would produce nearly continuous collisions. I 
have yet to find any reference to collision detection and/or 
collision avoidance on an ALE pilot channel. Does ALE provide some 
means of reducing contention? Without contention reduction, the 
practical number of simultaneous ALE stations sounding the same 
frequency at the recommended rates would be in the 30 to 60 range.

How could 1000 amateur ALE operators to be simultaneously QRV with 
one pilot channel per amateur band?

73,

Dave, AA6YQ

--- In digitalradio@yahoogroups.com, expeditionradio 
[EMAIL PROTECTED] wrote:

  John VE5MU wrote:
  If we have 1000 Ale stations sounding 24/7, how much QRM 
  will this create?  
 
 Hi John,
 
 It would be far less QRM than the average RTTY contest, such as we 
had
 this weekend that took over a large chunk of the ham bands with
 soundings. 
 
 In fact, it is unlikely that you would notice 1000 ALE operators on
 the air, unless you tune your VFO to the specific frequency the ALE
 operators are using. 
  
 In amateur radio ALE, there is only one pilot channel per ham band
 where repetitive sounding (station ID) happens on a regular basis. 
The
 nature of the way ALE works enables many stations to dynamically use
 the same channel on a time/space shared basis for various purposes,
 such as messaging, calling, sounding, and geo-position reporting. 
The
 global or regional capacity of a single channel for ALE is rather
 large. One channel is probabably enough to handle a 1000% increase 
in
 amateur radio ALE use over the next 5 years, if and when it becomes
 that popular. 
 
 It would be wonderful if we had 1000 ALE stations on the air 24/7.
 Perhaps the ALE On-The-AIR Week event in October will give us some
 idea of what is possible with a few hundred stations on at the same
 time. We don't really know yet, since this will be the first such 
event.
 
 Bonnie KQ6XA








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[digitalradio] Re: New to Digital HF -- PACTOR setup and hardware maybe needed???

2006-08-28 Thread Dave Bernstein
There are variants of Linux with pre-emptive scheduling; this enables 
guaranteed real-time response. Linux-based cellphones use this 
approach, for example.

See http://en.wikipedia.org/wiki/Hard_real-time

Dave, AA6YQ

--- In digitalradio@yahoogroups.com, Chris Jewell ae6vw-
[EMAIL PROTECTED] wrote:

 [EMAIL PROTECTED] writes:
   On Sun, 27 Aug 2006 22:02:33 -0500, John Becker [EMAIL PROTECTED] 
wrote:
   
   It was on a linux system
   But that does not matter.
   The problem is EVERY time the computer thinks
   what do I need to do now - the timing is lost and so
   is the link.
   
   ???
   now, I am not a geek for computers, but my Perl mobo has a pair 
of 3.1ghz cpus
   running with huge cache, and 2gb of ram.  the soundcard is 
running along with
   a pretty wide gateway and its own gb or so of ram.
   
   i just don't think that the wait states, if there are any, are 
going to be
   sufficient to bog down a real, damned slow modem since most of 
those are
   probably running at 1200 to 2400bps.  which is real slow.
   I have a dsl which supposedly is running at 3.2kbips but it is 
uploading
   slower than my pc is pushing it up the line.
   
   I am definitely confused which is NOT unusual in the least.
 
 Suppose you're using your sound card as a modem to receive Pactor I
 data.  Your sound card takes care of turning tones from the receiver
 into 1s and 0s.  There's no problem there.
 
 However, when a packet is finished arriving (and getting turned from
 frequency shifts into bytes of data), the application program 
running
 on the PC must verify the check, turn the radio around to transmit,
 and send either a positive or negative acknowledgement, then turn it
 back to receive mode so it can hear the next packet.
 
 The problem is that the OS may have dispatched some other process at
 the time, and the process that does the checking and sending of the
 ACK or NAK may not get a time slice from the OS soon enough to meet
 the timing requirements of the Pactor acknowledgement.
 
 If you were using a TNC, the processor in the TNC would be old and
 slow by current standards, but it would have nothing else to do
 EXCEPT check the checksum and send the ack or nak, while the much
 faster CPU in your Linux (or even worse, Windows) PC may be busy 
doing
 something else at the crucial time, since it is running a
 multitasking, and even potentially multi-user, OS.
 
 In principle, it seems that it should be possible to manage that
 problem in the Linux environment by:
 
 1.  adjusting the dispatching code in the OS so that it gives very
 short timeslices to processes, so the Pactor process can get
 CPU time sooner; and
 
 2.  running the Pactor process with a high dispatching priority
 (which the superuser can accomplish with a negative priority 
value
 on the renice command.)
 
 Shortening the time slice means that more of your CPU power is used 
up
 on trips through the dispatcher portion of the OS, rather than in
 running whatever user-mode application programs are ready to run, 
but
 with a fast computer that doesn't have a lot to do, that should be
 okay.
 
 In FreeBSD, you would issue a command like this as root:
# sysctl kern.sched.quantum=1
 The quantum is measured in microseconds.  The default value is 
100,000,
 or 100 milliseconds.  By changing it to 10,000, or 10 milliseconds, 
you
 make it possible for the highest-priority task to get use of the CPU
 within 10 milliseconds, rather than 100.
 
 I'm not really a Linux guy, so I don't know whether there is a
 comparable sysctl variable in Linux, or whether you can build a 
custom
 kernel that uses a shorter scheduling quantum, or what.
 
 --
 73 DE AE6VW, Chris Jewell  [EMAIL PROTECTED]  Gualala CA USA








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Re: [digitalradio] Re: New to Digital HF -- PACTOR setup and hardware maybe needed???

2006-08-28 Thread Steve Hajducek

GM Dave,

Yes, a technical item up for discussion.

I must assume that you have never done any Near Real Time Systems 
development such as ATE or Industrial Control applications under MS-Windows?

I on the other hand have and the WIN32 API beginning all the way back 
with Windows NT implemented a number of functions for embedded 
applications with critical timing requirements.

For example you have the GetTickCount() API call which has a 
resolution of 10ms and the QueryPerformanceCounter() which returns 
the resolution of a high-resolution performance counter to 0.8 
microseconds and then there are Spinlocks to synchronize timing 
events during an interrupt response or other similar activity, 
together with Process Priority and Asynchronous I/O and some other 
functions you can achieve nanosecond accuracy and make your 
application own the operating system to prevent other system 
processes from interfering with your critical timing needs.

It is my opinion that ARQ protocols such as AMTOR and others with 
their short ACK/NAK can implemented under MS-Windows 2000 
Professional and above taking the above approach.

/s/ Steve, N2CKH/AAR2EY


At 10:23 PM 8/27/2006, you wrote:
The issue is control over the operating system's scheduling
decisions. There are real-time versions of Linux that are comparable
in this dimension to the firmware running in a TNC; given sufficient
CPU horsepower, a Pactor-2 or Pactor-3 implementation on realtime
Linux is feasible.

The Windows scheduler cannot be adequately controlled for use timing-
critical applications; as a result, the response time requirements of
protocols like Pactor-2 and Pactor-3 cannot be guaranteed, no matter
how fast the CPU. Generating CW with consistent timing is a challenge
on this platform.

 73,

 Dave, AA6YQ




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Re: [digitalradio] Re: ALE QRM is minimal

2006-08-28 Thread Steve Hajducek

Hi Dave,

 At 10:53 PM 8/27/2006, you wrote:
 Does ALE provide some means of reducing contention?

I recommend that to answer all of your technical questions on subject 
ALE that you refer the actual Federal, Military and STANAG Standards 
which you can find on the Internet quite easily. You can start with a 
number of them at the following URL: 
http://www.n2ckh.com/MARS_ALE_FORUM/tecref.html

Listed below are the ALE Operational Rules taken directly from 
MIL-STD-188-141B APPENDIX A, take the time to read this and do 
additional research WRT the details of the referenced items herein 
and you should be satisfied that ALE is the most courteous digital 
mode with automatic operation you could ever want to see, compared to 
any other system that has ever been used on the Amateur Radio bands.

/s/ Steve, N2CKH/AAR2EY

A.4.4 ALE operational rules.
The ALE system shall incorporate the basic operational rules listed 
in table A-V. Some of these
rules may not be applicable in certain applications. For example, 
always listening is not
possible while transmitting with a transceiver or when using a common 
antenna with a separate
transmitter and receiver.

TABLE A-V. ALE operational rules.
1) Independent ALE receive capability (in parallel with other modems 
and simular audio receivers) (critical).
2) Always listening (for ALE signals) (critical).
3) Always will respond (unless deliberately inhibited).
4) Always scanning (if not otherwise in use).
5) Will not interfere with active channel carrying detectable traffic 
in accordance with table A-I (unless this
listen call function is overriden by the operator or other controller).
6) Always will exchange LQA with other stations when requested 
(unless inhibited), and always measures the
signal quality of others.
7) Will respond in the appropriate time slot to calls requiring 
slotted responses.
8) Always seek (unless inhibited) and maintain track of their 
connectivities with others.
9) Linking ALE stations employ highest mutual level of capability.
10) Minimize transmit and receive time on channel.
11) Automatically minimize power used (if capable).
NOTE : Listed in order of precedence.

TABLE A-I. Occupancy detection probability (2G and 3G).

WaveformSNR (dB in 3 kHz) Dwell Time (s) Detection Prob

ALE 0   2.0 0.80
 6   2.0 0.99

SSB Voice   6   2.0 0.80
 9   2.0 0.99

MIL-STD-188-110 0   2.0 0.80
(Serial Tone PSK)   6   2.0 0.99

STANAG  45290   2.0 0.80
 6   2.0 0.99

STANAG 4285 0   2.0 0.80
 6   2.0 0.99





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Re: [digitalradio] Re: ALE QRM is minimal

2006-08-28 Thread KV9U
Several key points on Bonnie's comments:

1) RTTY contests are human operating events. There is no automatic RTTY 
that I am aware of. Big difference! It is one thing to find an apparent 
hole to TX into, but are able to back off if it is busy. ALE would 
be nearly inoperative during a contest if it is truly monitoring the 
frequencies. Frequently, you will have wall to wall TX that take up the 
band.

If you ever think that you can make an argument to preempt human 
operation with machine, you will lose with the great majority of hams.

2) Having many ALE soundings on one frequency is beginning to sound like 
packet collision issues where nothing gets through. This can work for 
military, commercial, MARS, use perhaps where you have a fairly small 
number of stations that would send at any one time. But if you had even 
two stations sending on top of each other, wouldn't they trash the ALE 
packet?

Then imagine having a lot more than 2! And if you only sound once an 
hour, you won't even know that it did not work. Is this really practical 
for amateur radio use in a band that has long distance capabilities 
(sometimes world wide)?

73,

Rick, KV9U


expeditionradio wrote:


It would be far less QRM than the average RTTY contest, such as we had
this weekend that took over a large chunk of the ham bands with
soundings. 

In fact, it is unlikely that you would notice 1000 ALE operators on
the air, unless you tune your VFO to the specific frequency the ALE
operators are using. 
 
In amateur radio ALE, there is only one pilot channel per ham band
where repetitive sounding (station ID) happens on a regular basis. The
nature of the way ALE works enables many stations to dynamically use
the same channel on a time/space shared basis for various purposes,
such as messaging, calling, sounding, and geo-position reporting. The
global or regional capacity of a single channel for ALE is rather
large. One channel is probabably enough to handle a 1000% increase in
amateur radio ALE use over the next 5 years, if and when it becomes
that popular. 

It would be wonderful if we had 1000 ALE stations on the air 24/7.
Perhaps the ALE On-The-AIR Week event in October will give us some
idea of what is possible with a few hundred stations on at the same
time. We don't really know yet, since this will be the first such event.

Bonnie KQ6XA
 

  




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[digitalradio] Timing requirements of digital ARQ

2006-08-28 Thread KV9U
Since I am not a programmer, other than taking some rudimentary courses, 
reading some of Don Lancaster's books, and knowing that it is not 
something I could ever do very well, something still doesn't seem right 
to me when it comes to the claim that computers just can not meet the 
timing requirements of ARQ digital modes.

When I use the computer to key my ICOM 756 Pro 2 with Dave's DX Lab 
Commander software in order to run other programs that interface with 
Commander (such as Multipsk which is my main digital sound card 
program), it does not seem to have much latency at all. And if one could 
key relatively slow CW as is done with the software for the SDR1000, why 
would that not be almost a magnitude faster than you would need for 
adequate ARQ switching speed?

How many ms do you need?

Rick, KN6KB, the inventor of SCAMP, has said that the power in the 
typical Windows OS computer we use is something like a magnitude less 
than that available in the dedicated SCS modems and that is why they 
perform so well compared to a computer (for the dedicated part). I 
wonder where the dividing line will come so that computers will at least 
match the SCS type units?

73,

Rick, KV9U


Dave Bernstein wrote:

The issue is control over the operating system's scheduling 
decisions. There are real-time versions of Linux that are comparable 
in this dimension to the firmware running in a TNC; given sufficient 
CPU horsepower, a Pactor-2 or Pactor-3 implementation on realtime 
Linux is feasible.

The Windows scheduler cannot be adequately controlled for use timing-
critical applications; as a result, the response time requirements of 
protocols like Pactor-2 and Pactor-3 cannot be guaranteed, no matter 
how fast the CPU. Generating CW with consistent timing is a challenge 
on this platform.

73,

Dave, AA6YQ

  




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Re: [digitalradio] New to Digital HF -- PACTOR setup and hardware maybe needed???

2006-08-28 Thread KV9U
Chris,

What is your view on using pipelined programming such as what was used 
in the SCAMP mode to get around this issue with moving the ACK to the 
next packet. The main penalty is latency for the user, but it seems 
manageable.

73,

Rick, KV9U


Chris Jewell wrote:

Suppose you're using your sound card as a modem to receive Pactor I
data.  Your sound card takes care of turning tones from the receiver
into 1s and 0s.  There's no problem there.

However, when a packet is finished arriving (and getting turned from
frequency shifts into bytes of data), the application program running
on the PC must verify the check, turn the radio around to transmit,
and send either a positive or negative acknowledgement, then turn it
back to receive mode so it can hear the next packet.

The problem is that the OS may have dispatched some other process at
the time, and the process that does the checking and sending of the
ACK or NAK may not get a time slice from the OS soon enough to meet
the timing requirements of the Pactor acknowledgement.

If you were using a TNC, the processor in the TNC would be old and
slow by current standards, but it would have nothing else to do
EXCEPT check the checksum and send the ack or nak, while the much
faster CPU in your Linux (or even worse, Windows) PC may be busy doing
something else at the crucial time, since it is running a
multitasking, and even potentially multi-user, OS.

In principle, it seems that it should be possible to manage that
problem in the Linux environment by:

1.  adjusting the dispatching code in the OS so that it gives very
short timeslices to processes, so the Pactor process can get
CPU time sooner; and

2.  running the Pactor process with a high dispatching priority
(which the superuser can accomplish with a negative priority value
on the renice command.)

Shortening the time slice means that more of your CPU power is used up
on trips through the dispatcher portion of the OS, rather than in
running whatever user-mode application programs are ready to run, but
with a fast computer that doesn't have a lot to do, that should be
okay.

In FreeBSD, you would issue a command like this as root:
   # sysctl kern.sched.quantum=1
The quantum is measured in microseconds.  The default value is 100,000,
or 100 milliseconds.  By changing it to 10,000, or 10 milliseconds, you
make it possible for the highest-priority task to get use of the CPU
within 10 milliseconds, rather than 100.

I'm not really a Linux guy, so I don't know whether there is a
comparable sysctl variable in Linux, or whether you can build a custom
kernel that uses a shorter scheduling quantum, or what.

--
73 DE AE6VW, Chris Jewell  [EMAIL PROTECTED]  Gualala CA USA

  




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[digitalradio] Re: Timing requirements of digital ARQ

2006-08-28 Thread mulveyraa2
--- In digitalradio@yahoogroups.com, KV9U [EMAIL PROTECTED] wrote:

 Since I am not a programmer, other than taking some rudimentary 
courses, 
 reading some of Don Lancaster's books, and knowing that it is not 
 something I could ever do very well, something still doesn't seem 
right 
 to me when it comes to the claim that computers just can not meet 
the 
 timing requirements of ARQ digital modes.
 
 When I use the computer to key my ICOM 756 Pro 2 with Dave's DX Lab 
 Commander software in order to run other programs that interface 
with 
 Commander (such as Multipsk which is my main digital sound card 
 program), it does not seem to have much latency at all. And if one 
could 
 key relatively slow CW as is done with the software for the 
SDR1000, why 
 would that not be almost a magnitude faster than you would need for 
 adequate ARQ switching speed?
 
 How many ms do you need?
 
 Rick, KN6KB, the inventor of SCAMP, has said that the power in the 
 typical Windows OS computer we use is something like a magnitude 
less 
 than that available in the dedicated SCS modems and that is why 
they 
 perform so well compared to a computer (for the dedicated part). I 
 wonder where the dividing line will come so that computers will at 
least 
 match the SCS type units?
 


The problem is not that the PC's are not fast enough, it's that the 
timing of events is not predictable enough.  In even the smallest 
microcontrollers, you can control the timing of events and processing 
down to a microsecond, if you so choose.  For example, you can set 
the state of a pin, wait 8.76 milliseconds, change the state - and 
you KNOW that the state will be changed at that 8.76ms point, no 
if's, and's, or but's.  In a typical PC OS, however, you lose that 
predictability.  You can tell the OS, I want to wait 8.76ms, so you 
can then go and change the state of the pin, but, in reality, you may 
not actually be able to do it for 8, 10, or 30ms later, because the 
OS is busy doing other things, in the meantime.  Asking for an 8.76ms 
pause doesn't mean you'll GET an 8.76ms pause.

   I can't speak as to the accuracy of the posting that indicated 
that Windows has some facilities to accomodate real-time 
programming.  If that's the case, I have to wonder why nobody has 
done it under Pactor I, at least, for Windows.  Years ago, there were 
P-I *decoders* that worked under DOS.  And now there's a P-I 
application for Linux called 'hfterm'.  But, I've compared on-the-air 
performance between hfterm and my SCC PTC-II modem in P-I mode, and 
there's just no comparison - the SCS blows hfterm away.  At least, so 
far.  ;-)

- Rich







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[digitalradio] Re: New to Digital HF -- PACTOR setup and hardware maybe needed???

2006-08-28 Thread Dave Bernstein
Precise timing isn't the issue, Steve. WinWarbler originally used 
GetTickCount() and QueryPerformanceCounter() in its CW generation 
code, but a high-resolution timer using the multimedia library is 
sufficiently accurate and more convenient. The problem is thread 
scheduling. WinWarbler uses SetPriorityClass to establish 
REALTIME_PRIORITY_CLASS and SetThreadPriority to establish 
THREAD_PRIORITY_TIME_CRITICAL . Nonetheless, there are many CPU-
consumptive events that Windows insists on handling at higher 
priority -- such as displaying and moving the Task Manager window. 
Spin locks are not an acceptable solution IMHO; the user would find 
it disconcerting if the mouse pointer stopped following mouse 
movements during CW generation or Pactor-3 operation, for example. 
Yes, one could dedicate a headless Windows PC to executing one 
application like Pactor-3, but what would be the point? Even the most 
expensive Pactor-3 TNC would be cheaper and more compact to boot.

I didn't comment on the feasibility of implementing AMTOR on Windows; 
Pactor-2 and Pactor-3 were the protocols I mentioned.

From my days as a CPU architect and designer, I have lots of 
experience with real-time operating systems and applications -- but 
WinWarbler was my first encounter with extracting real-time 
performance from Windows. Any suggestions or pointers you have in 
this area would be appreciated.

73,

 Dvae, AA6YQ


--- In digitalradio@yahoogroups.com, Steve Hajducek [EMAIL PROTECTED] 
wrote:

 
 GM Dave,
 
 Yes, a technical item up for discussion.
 
 I must assume that you have never done any Near Real Time Systems 
 development such as ATE or Industrial Control applications under MS-
Windows?
 
 I on the other hand have and the WIN32 API beginning all the way 
back 
 with Windows NT implemented a number of functions for embedded 
 applications with critical timing requirements.
 
 For example you have the GetTickCount() API call which has a 
 resolution of 10ms and the QueryPerformanceCounter() which returns 
 the resolution of a high-resolution performance counter to 0.8 
 microseconds and then there are Spinlocks to synchronize timing 
 events during an interrupt response or other similar activity, 
 together with Process Priority and Asynchronous I/O and some other 
 functions you can achieve nanosecond accuracy and make your 
 application own the operating system to prevent other system 
 processes from interfering with your critical timing needs.
 
 It is my opinion that ARQ protocols such as AMTOR and others with 
 their short ACK/NAK can implemented under MS-Windows 2000 
 Professional and above taking the above approach.
 
 /s/ Steve, N2CKH/AAR2EY
 
 
 At 10:23 PM 8/27/2006, you wrote:
 The issue is control over the operating system's scheduling
 decisions. There are real-time versions of Linux that are 
comparable
 in this dimension to the firmware running in a TNC; given 
sufficient
 CPU horsepower, a Pactor-2 or Pactor-3 implementation on realtime
 Linux is feasible.
 
 The Windows scheduler cannot be adequately controlled for use 
timing-
 critical applications; as a result, the response time requirements 
of
 protocols like Pactor-2 and Pactor-3 cannot be guaranteed, no 
matter
 how fast the CPU. Generating CW with consistent timing is a 
challenge
 on this platform.
 
  73,
 
  Dave, AA6YQ







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[digitalradio] Re: ALE QRM is minimal

2006-08-28 Thread Dave Bernstein
Steve, I asked a few simple questions about the amateur 
implementation of ALE; these questions were not focused on 
politeness, but rather on understanding how many ALE users can be 
simultaneously QRV if there's one pilot channel per amateur band. 
Bonnie claimed 1000, but two multiplications and a division yields a 
much lower number. If the model underlying this math is incorrect, 
please set me straight.

I have reviewed enough of the military documentation to understand 
that they employ dedicated ALE transceivers capable of much faster 
scanning rates. As a result, sounding duration is signficantly 
reduced, and channel capacity increases in proportion. But one ham 
with an amateur transceiver limited to a 2 channel-per-second scan 
rate would force all ALE participants to sound for 20 seconds, even 
if their equipment could scan more rapidly. Do I have this right?

Since the 20 second sounding time is calculated to allow each station 
to scan 40 frequencies without missing a sounding, one obvious step 
to increase capacity would be to reduce the number of frequencies 
being scanned. If there's only one pilot channel per amateur band, 
scanning could be reduced to 10 frequencies, which would allow 
sounding time to be reduced to 5 seconds. This would increase 
capacity by a factor of 4 -- to ~120 simultaneous users, if I 
understand correctly.

73,

Dave, AA6YQ


--- In digitalradio@yahoogroups.com, Steve Hajducek [EMAIL PROTECTED] 
wrote:

 
 Hi Dave,
 
  At 10:53 PM 8/27/2006, you wrote:
  Does ALE provide some means of reducing contention?
 
 I recommend that to answer all of your technical questions on 
subject 
 ALE that you refer the actual Federal, Military and STANAG 
Standards 
 which you can find on the Internet quite easily. You can start with 
a 
 number of them at the following URL: 
 http://www.n2ckh.com/MARS_ALE_FORUM/tecref.html
 
 Listed below are the ALE Operational Rules taken directly from 
 MIL-STD-188-141B APPENDIX A, take the time to read this and do 
 additional research WRT the details of the referenced items herein 
 and you should be satisfied that ALE is the most courteous digital 
 mode with automatic operation you could ever want to see, compared 
to 
 any other system that has ever been used on the Amateur Radio bands.
 
 /s/ Steve, N2CKH/AAR2EY
 
 A.4.4 ALE operational rules.
 The ALE system shall incorporate the basic operational rules listed 
 in table A-V. Some of these
 rules may not be applicable in certain applications. For example, 
 always listening is not
 possible while transmitting with a transceiver or when using a 
common 
 antenna with a separate
 transmitter and receiver.
 
 TABLE A-V. ALE operational rules.
 1) Independent ALE receive capability (in parallel with other 
modems 
 and simular audio receivers) (critical).
 2) Always listening (for ALE signals) (critical).
 3) Always will respond (unless deliberately inhibited).
 4) Always scanning (if not otherwise in use).
 5) Will not interfere with active channel carrying detectable 
traffic 
 in accordance with table A-I (unless this
 listen call function is overriden by the operator or other 
controller).
 6) Always will exchange LQA with other stations when requested 
 (unless inhibited), and always measures the
 signal quality of others.
 7) Will respond in the appropriate time slot to calls requiring 
 slotted responses.
 8) Always seek (unless inhibited) and maintain track of their 
 connectivities with others.
 9) Linking ALE stations employ highest mutual level of capability.
 10) Minimize transmit and receive time on channel.
 11) Automatically minimize power used (if capable).
 NOTE : Listed in order of precedence.
 
 TABLE A-I. Occupancy detection probability (2G and 3G).
 
 WaveformSNR (dB in 3 kHz) Dwell Time (s) Detection Prob
 
 ALE 0   2.0 0.80
  6   2.0 0.99
 
 SSB Voice   6   2.0 0.80
  9   2.0 0.99
 
 MIL-STD-188-110 0   2.0 0.80
 (Serial Tone PSK)   6   2.0 0.99
 
 STANAG  45290   2.0 0.80
  6   2.0 0.99
 
 STANAG 4285 0   2.0 0.80
  6   2.0 0.99








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Re: [digitalradio] ALE QRM

2006-08-28 Thread John Champa
Yes!  Just select a calling frequency.  Then move off.


From: kd4e [EMAIL PROTECTED]
Reply-To: digitalradio@yahoogroups.com
To: digitalradio@yahoogroups.com
Subject: Re: [digitalradio] ALE QRM
Date: Sun, 27 Aug 2006 23:32:12 +0800

Someone somewhere will *have* to sound* else no one
anywhere will make a contact!  :-)

That said, in small numbers the ALE sounding-QRM is
unlikely to be unbearable.

Were the usage numbers to increase then the sounding-QRM
could definitely become more than an occasional nuisance.

Since ALE is mostly of value in military-like emergency
communications support there is no good argument for it
to be spread all over all of the Ham HF spectrum.  Tiny
slivers of each band answer the propagation availability
question.  Spinning the VFO and listening solves the
available frequency question.

Best of both worlds, least probability of QRM.

WDYT?  doc

  Where did you get the idea that we on ALE are sounding 24/7 ? ?
  If you check the HFLINK web site you will notice to the set up is
  asking that the system is *NOT* be sounding...
 
  please seehttp://www.hflink.com/beta/pcale1061options.jpg
 
  John, W0JAB
 
 
  At 09:30 PM 8/27/2006, you wrote:
  It is truly amazing to me that the same hams who have torn up this 
group
  time after time, objecting long and vigorously against
  automatic Pactor stations , are promoting the concept of ALE sounding 
on
  HF 24/7
 
  Talk about being hypocritical !!! If we have 1000 Ale stations sounding
  24/7, how much QRM will this create?
 
  John
  VE5MU



--

Thanks!  73,
doc, KD4E
... somewhere in FL
URL:  bibleseven (dot) com




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RE: [digitalradio] Re: Can You Call Another Ham On The Air? Right Now?

2006-08-28 Thread John Champa
Ahhh...we're a tiny minority.  Most Hams know only SSB and CW.

Besides, most of the new digital modes seems to come out of Europe!

From: Dave Bernstein [EMAIL PROTECTED]
Reply-To: digitalradio@yahoogroups.com
To: digitalradio@yahoogroups.com
Subject: [digitalradio] Re: Can You Call Another Ham On The Air? Right Now?
Date: Mon, 28 Aug 2006 05:59:07 -

If as you say, ham radio operators have not been thinking outside
the box, and are largely content with the status quo, having never
known anything better, then how do you explain

- the blizzard of new digital modes developed over the past 5 years

- the rapid adoption of panoramic reception and broadband decoding

- the use of software-defined transceivers

?

 73,

Dave, AA6YQ


--- In digitalradio@yahoogroups.com, expeditionradio
[EMAIL PROTECTED] wrote:
 
   Dave G0DJA David
   It's a bit of a silly arguement to say we should be able to
   call up someone who we probably have never spoken to before
   and have not got any idea whether they use the same bands
   and modes as we do...
 
  Hi Dave,
 
  I believe it is not silly at all to call specific hams we have never
  spoken with before, without knowing bands or modes they use...
simply
  by knowing their callsign.
 
  I suggest that the reason some might think it is odd, is that ham
  radio operators have not been thinking outside the box, and are
  largely content with the status quo, having never known anything
better.
 
  Calling each other on the air is technically feasible in many ways,
  especially with our available digital radio technologies,
  microprocessor-driven transceivers, and computers.
 
  In posing the question, I had hoped to spur some thought and
  intelligent discussion about different ways to dependably initiate
  communications with each other on the air. I'm interested in all
kinds
  of methods, especially the ones that do not require schedules,
manual
  net monitoring, or telephone calls.
 
  Bonnie KQ6XA
 
  .
 








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Re: [digitalradio] Re: ALE QRM is minimal

2006-08-28 Thread KV9U
One of the main interests that I have in digital modes is getting a 
message through the most difficult conditions, completely intact as 
sent, and as fast as possible. I was looking at the STANAG 5066 
specifications and test results, (Steve has some below), and quite 
frankly I am concerned that this standard has what I would normally 
consider to be unacceptable performance (non performance) with weak 
signals.

I am not sure what kind of cps or wpm throughput the bit rates mean but 
it I wonder how it compares to SCAMP running at 10 db S/N?  Because 
SCAMP only operated down to about +10 db S/N (maybe slightly better), it 
was rejected as unacceptable for practical messaging.

 From the info on Steve's site:

http://www.n2ckh.com/MARS_ALE_FORUM/MIL-STD-188-110B.pdf

Here are some claimed performance levels:

Bit rate  Multipath  SNR  BER

4800 2 ms 27 db 1 x e-3 with .5 Hz 
fading BW
2400 2 ms 30 db 1 x e-3 with  5 Hz 
fading BW
1200 2 ms 11 db 1 x e-5 with  1 Hz 
fading BW
300   5 ms   7 db 1 x e-5 with  5 Hz 
fading BW
75 5 ms   2 db 1 x e-5 with  5 
Hz fading BW

Even with the slowest 75 bps, and a multipath of 5 ms, it can only work 
down to 2 db ABOVE the noise! This is not good. From personal 
experience, it is not easy to get even 10 db S/N signals with typical 
amateur signals with modest antennas on the lower bands.

They even show some constellations at 64 QAM. From what the SSTV folks 
have said, 64 QAM is not really a useful mode on HF. Perhaps that is 
because they are not using ARQ?

Note also that the multipaths are moderate to low compared to worst case 
HF propagation. I question whether this stuff can work under many 
conditions we routinely operate with sound card modes (but are not 100% 
copy without ARQ).

The BER that this system can handle seems to indicate that the channel 
has to be rather good.  These BER's seem to be more appropriate for what 
we would expect on equipment designed for VHF and up ...  aren't they?

For those of you who have used STANAG 5066 waveforms, what kind of 
throughput have you experienced with real world connections?

The deeper I examine this NATO standardized agreeement, the more it is 
beginning to look like another one of those the emperor has no clothes 
findings.

Thanks and 73,

Rick, KV9U



Steve Hajducek wrote:

I recommend that to answer all of your technical questions on subject 
ALE that you refer the actual Federal, Military and STANAG Standards 
which you can find on the Internet quite easily. You can start with a 
number of them at the following URL: 
http://www.n2ckh.com/MARS_ALE_FORUM/tecref.html

Listed below are the ALE Operational Rules taken directly from 
MIL-STD-188-141B APPENDIX A, take the time to read this and do 
additional research WRT the details of the referenced items herein 
and you should be satisfied that ALE is the most courteous digital 
mode with automatic operation you could ever want to see, compared to 
any other system that has ever been used on the Amateur Radio bands.

/s/ Steve, N2CKH/AAR2EY

A.4.4 ALE operational rules.
The ALE system shall incorporate the basic operational rules listed 
in table A-V. Some of these
rules may not be applicable in certain applications. For example, 
always listening is not
possible while transmitting with a transceiver or when using a common 
antenna with a separate
transmitter and receiver.

TABLE A-V. ALE operational rules.
1) Independent ALE receive capability (in parallel with other modems 
and simular audio receivers) (critical).
2) Always listening (for ALE signals) (critical).
3) Always will respond (unless deliberately inhibited).
4) Always scanning (if not otherwise in use).
5) Will not interfere with active channel carrying detectable traffic 
in accordance with table A-I (unless this
listen call function is overriden by the operator or other controller).
6) Always will exchange LQA with other stations when requested 
(unless inhibited), and always measures the
signal quality of others.
7) Will respond in the appropriate time slot to calls requiring 
slotted responses.
8) Always seek (unless inhibited) and maintain track of their 
connectivities with others.
9) Linking ALE stations employ highest mutual level of capability.
10) Minimize transmit and receive time on channel.
11) Automatically minimize power used (if capable).
NOTE : Listed in order of precedence.

TABLE A-I. Occupancy detection probability (2G and 3G).

WaveformSNR (dB in 3 kHz) Dwell Time (s) Detection Prob

ALE 0   2.0 0.80
 6   2.0 0.99

SSB Voice   6   2.0 0.80
 9   2.0 0.99

MIL-STD-188-110 0   2.0 0.80

RE: [digitalradio] Re: The digital throughput challenge on H

2006-08-28 Thread DuBose Walt Civ AETC CONS/LGCA
New/updated Routing Information...station availibility, frequency changes, etc? 
 Currently in NTS called net bulletins.

73...K5YFW

-Original Message-

...under what circumstances would a message transport layer 
require one-to-many transmission?

   73,
  Dave, AA6YQ



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Re: [digitalradio] Re: ALE QRM is minimal

2006-08-28 Thread Steve Hajducek

Hi Dave,

At 10:46 AM 8/28/2006, you wrote:


I have reviewed enough of the military documentation to understand
that they employ dedicated ALE transceivers capable of much faster
scanning rates.

Really?  Please enlighten me, I was under the impression that the ALE 
scan rates of 1, 2 and 5 ch/sec was it at present and that the future 
goal as stated in MIL-STD-188-141B was 10 ch/sec. The PC-ALE software 
supports 1, 2 and 5 ch/sec with an HF transceiver that is cable of 
all selections.


  As a result, sounding duration is signficantly
reduced,

Sorry, but you will have to explain to me how Sounding duration 
decreases with an increase in the Scan Rate.

  and channel capacity increases in proportion.

Well not exactly. A 2 ch/sec scan rate allows you to cover the same 
number of channels faster than a 1 ch/sec scan rate and thus increase 
the odds of hearing a Sounding or a Linking call sooner.

Running 2 or 5 ch/sec will also permit the station to be part of more 
than one ALE network at the same time, not an issue per see with 
Amateur Radio, but if two networks had scan groups of 10 channels 
each, you could scan both with excellent results.

The number of channels you scan does have an effect on your 
Soundings, you sound longer when you have more channels in the mix. 
There are variable here as we are now at a stage were you have 3 
generations of ALE. The latest ALE technology supports GPS time 
synchronization of the Scanning/Sounding which in the future will 
radically reduce BER/SNR data transfer for LQA ranking when all 
user's can support it.


  But one ham
with an amateur transceiver limited to a 2 channel-per-second scan
rate would force all ALE participants to sound for 20 seconds, even
if their equipment could scan more rapidly. Do I have this right?

The details are to be found in MIL-STD-188-141B Appendix A. 
Regardless of the scan rate, if the controller Sounds based on number 
of channels in the scan group its less than 1 second per channel. The 
minimum redundant sound length (Trs) is equal to the standard 
one-word address leading call; that is, Trs = Tlc min = 2 Ta min = 2 
Trw = 784 ms. Thus for 12 channels it would be about 9.4 seconds 
depending on your address length being sent. The address length is 
based on an ALE Word which is 3 ASCII characters, for Amateur Radio 
applications we would being using 2 ALE Words as there are no 3 
character callsigns, whereas in the Military and Government world 
there are 3 character ALE Self Addresses being used. So W1AW, N2CKH 
and WB2XYZ are all 2 ALE Words were automatic padding is used to fill 
the second word. The least number of ALE Words the more efficient and 
reliable is the system. One would now want to use WB2XYZ/W6 to 
indicate they are in California. For AQC-ALE where many things were 
changed to make things even more efficient, a 2 ALE Word is the 
maximum allowed, whereas the original ALE allowed a 5 ALE Word (15 
character) Address to support the Military Automatic Digital Network 
(AUTODIN) system to directly link a Phone Patch call. Feel free to 
double check me with the standards, I am no expert on all this stuff 
and I am not perfect either, I make calculation errors often.

/s/ Steve, N2CKH




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RE: [digitalradio] Re: ALE QRM is minimal

2006-08-28 Thread rattray
Hello Bonnie - what is ALE please? - 73 Bruce.

72/73 - Bruce ve5rc/ve5qrp - QRP-C#1, QRP-L#886, A1 Operator
Enter QRP-Canada's RUN with RAC contest -
    details - http://www.qrp-canada.com 
 



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Re: [digitalradio] Re: New to Digital HF -- PACTOR setup and hardware maybe needed???

2006-08-28 Thread Steve Hajducek

Hi Dave,

I mentioned AMTOR as its timing is more robust that PACTOR I.

As I have stated to the MARS-ALE users, the future version of that 
tool when PACTOR I support is added ont he PCSDM will pretty much own 
the OS, not a problem for our purposes as that one program running is 
our only focus. GTOR is less demanding, but may be require the same 
approach as PACTOR I. This is not required of DBM ARQ which is much 
less demanding that GTOR.

/s/ Steve, N2CKH

At 06:43 AM 8/28/2006, you wrote:
Precise timing isn't the issue, Steve. WinWarbler originally used
GetTickCount() and QueryPerformanceCounter() in its CW generation
code, but a high-resolution timer using the multimedia library is
sufficiently accurate and more convenient. The problem is thread
scheduling. WinWarbler uses SetPriorityClass to establish
REALTIME_PRIORITY_CLASS and SetThreadPriority to establish
THREAD_PRIORITY_TIME_CRITICAL . Nonetheless, there are many CPU-
consumptive events that Windows insists on handling at higher
priority -- such as displaying and moving the Task Manager window.
Spin locks are not an acceptable solution IMHO; the user would find
it disconcerting if the mouse pointer stopped following mouse
movements during CW generation or Pactor-3 operation, for example.
Yes, one could dedicate a headless Windows PC to executing one
application like Pactor-3, but what would be the point? Even the most
expensive Pactor-3 TNC would be cheaper and more compact to boot.

I didn't comment on the feasibility of implementing AMTOR on Windows;
Pactor-2 and Pactor-3 were the protocols I mentioned.

 From my days as a CPU architect and designer, I have lots of
experience with real-time operating systems and applications -- but
WinWarbler was my first encounter with extracting real-time
performance from Windows. Any suggestions or pointers you have in
this area would be appreciated.

 73,

  Dvae, AA6YQ




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[digitalradio] Re: Timing requirements of digital ARQ

2006-08-28 Thread Dave Bernstein
To use your example, Rick, if Windows introduces a 10 ms delay from 
the time when you strike a button to initiate transmission in 
MultiPSK until the point where Commander sends a CI-V Transmit! 
command to your 756 Pro2, you'd never notice. However, such a delay 
in confirming reception of a Pactor-3 packet would break the protocol.

The SCS PTC-II Pro, which supports both Pactor-2 and Pactor-3, 
utilizes a Motorola (now Freescale) 68360 clocked at 25 MHz and a 
56303 24-bit DSP clocked at 100 MHz; see

http://www.globalmarinenet.net/ptc.htm

The 68360 is a descendant of the venerable 68000; its probably being 
used for administrative tasks like running the serial port and 
command line interface. The 56303 DSP does the heavy lifting. While 
its DSP instruction set is advantageous, its performance is modest by 
current standards; see

http://www.bdti.com/articles/benchmark_icspat99.pdf#search=%22dsp%
20benchmarks%22

I am unable to locate a benchmark comparing the performance of 
dedicated DSP chips like the 56303 DSP with general purpose 
microprocessors, but I suspect that a 1 GHz Pentium could handle the 
processing load.

   73,

   Dave, AA6YQ


--- In digitalradio@yahoogroups.com, KV9U [EMAIL PROTECTED] wrote:

 Since I am not a programmer, other than taking some rudimentary 
courses, 
 reading some of Don Lancaster's books, and knowing that it is not 
 something I could ever do very well, something still doesn't seem 
right 
 to me when it comes to the claim that computers just can not meet 
the 
 timing requirements of ARQ digital modes.
 
 When I use the computer to key my ICOM 756 Pro 2 with Dave's DX Lab 
 Commander software in order to run other programs that interface 
with 
 Commander (such as Multipsk which is my main digital sound card 
 program), it does not seem to have much latency at all. And if one 
could 
 key relatively slow CW as is done with the software for the 
SDR1000, why 
 would that not be almost a magnitude faster than you would need for 
 adequate ARQ switching speed?
 
 How many ms do you need?
 
 Rick, KN6KB, the inventor of SCAMP, has said that the power in the 
 typical Windows OS computer we use is something like a magnitude 
less 
 than that available in the dedicated SCS modems and that is why 
they 
 perform so well compared to a computer (for the dedicated part). I 
 wonder where the dividing line will come so that computers will at 
least 
 match the SCS type units?
 
 73,
 
 Rick, KV9U
 
 
 Dave Bernstein wrote:
 
 The issue is control over the operating system's scheduling 
 decisions. There are real-time versions of Linux that are 
comparable 
 in this dimension to the firmware running in a TNC; given 
sufficient 
 CPU horsepower, a Pactor-2 or Pactor-3 implementation on realtime 
 Linux is feasible.
 
 The Windows scheduler cannot be adequately controlled for use 
timing-
 critical applications; as a result, the response time requirements 
of 
 protocols like Pactor-2 and Pactor-3 cannot be guaranteed, no 
matter 
 how fast the CPU. Generating CW with consistent timing is a 
challenge 
 on this platform.
 
 73,
 
 Dave, AA6YQ
 
   
 








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[digitalradio] Re: The digital throughput challenge on H

2006-08-28 Thread Dave Bernstein
Those are all low-occurrence events that could be implemented with 
one-to-one messages with no significant performance degradation.

One-to-one messaging with ARQ would seem optimal. KISS, remember?

   73,

   Dave, AA6YQ


--- In digitalradio@yahoogroups.com, DuBose Walt Civ AETC CONS/LGCA 
[EMAIL PROTECTED] wrote:

 New/updated Routing Information...station availibility, frequency 
changes, etc?  Currently in NTS called net bulletins.
 
 73...K5YFW
 
 -Original Message-
 
 ...under what circumstances would a message transport layer 
 require one-to-many transmission?
 
73,
   Dave, AA6YQ








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Re: [digitalradio] Re: ALE QRM is minimal

2006-08-28 Thread Steve Hajducek

Hi Rick,

Just time for a quick comment.

Don't confuse STANAG 5066 Data Link Protocol (DLP) as covered in 
MIL-STD-188-141B which is a Data Link Protocol at the Physical Layer 
with STANAG 5066 which is a network protocol at the Link Layer. 
Basically and DLP with the need ARQ support and speed can be used at 
the Physical Layer. If an MT-63 Adaptive ARQ protocol with a 
transport layer and enough speed were to develop it could be used.

STANAG 5066 DLP (S5066) replaced FED-STD-1052 DLP (FS-1052) going 
from MIL-STD-188-141A to MIL-STD-188-141B. Both are DLP's that make 
use of the MIL-STD-188-110x modems, both provide and ARQ protocol, 
where 5066 DLP is much improved.

We use FS-1052 daily in MARS, we get full 2400bps throughput on a 
good channel with stations that are properly configured. We have not 
yet implemented S5066, its on the To Do list.

/s/ Steve, N2CKH/AAR2EY

At 11:16 AM 8/28/2006, you wrote:
One of the main interests that I have in digital modes is getting a
message through the most difficult conditions, completely intact as
sent, and as fast as possible. I was looking at the STANAG 5066
specifications and test results, (Steve has some below), and quite
frankly I am concerned that this standard has what I would normally
consider to be unacceptable performance (non performance) with weak
signals.

I am not sure what kind of cps or wpm throughput the bit rates mean but
it I wonder how it compares to SCAMP running at 10 db S/N?  Because
SCAMP only operated down to about +10 db S/N (maybe slightly better), it
was rejected as unacceptable for practical messaging.

  From the info on Steve's site:

http://www.n2ckh.com/MARS_ALE_FORUM/MIL-STD-188-110B.pdf

Here are some claimed performance levels:

Bit rate  Multipath  SNR  BER

4800 2 ms 27 db 1 x e-3 with .5 Hz
fading BW
2400 2 ms 30 db 1 x e-3 with  5 Hz
fading BW
1200 2 ms 11 db 1 x e-5 with  1 Hz
fading BW
300   5 ms   7 db 1 x e-5 with  5 Hz
fading BW
75 5 ms   2 db 1 x e-5 with  5
Hz fading BW

Even with the slowest 75 bps, and a multipath of 5 ms, it can only work
down to 2 db ABOVE the noise! This is not good. From personal
experience, it is not easy to get even 10 db S/N signals with typical
amateur signals with modest antennas on the lower bands.

They even show some constellations at 64 QAM. From what the SSTV folks
have said, 64 QAM is not really a useful mode on HF. Perhaps that is
because they are not using ARQ?

Note also that the multipaths are moderate to low compared to worst case
HF propagation. I question whether this stuff can work under many
conditions we routinely operate with sound card modes (but are not 100%
copy without ARQ).

The BER that this system can handle seems to indicate that the channel
has to be rather good.  These BER's seem to be more appropriate for what
we would expect on equipment designed for VHF and up ...  aren't they?

For those of you who have used STANAG 5066 waveforms, what kind of
throughput have you experienced with real world connections?

The deeper I examine this NATO standardized agreeement, the more it is
beginning to look like another one of those the emperor has no clothes
findings.

Thanks and 73,

Rick, KV9U



Steve Hajducek wrote:

 I recommend that to answer all of your technical questions on subject
 ALE that you refer the actual Federal, Military and STANAG Standards
 which you can find on the Internet quite easily. You can start with a
 number of them at the following URL:
 http://www.n2ckh.com/MARS_ALE_FORUM/tecref.html
 
 Listed below are the ALE Operational Rules taken directly from
 MIL-STD-188-141B APPENDIX A, take the time to read this and do
 additional research WRT the details of the referenced items herein
 and you should be satisfied that ALE is the most courteous digital
 mode with automatic operation you could ever want to see, compared to
 any other system that has ever been used on the Amateur Radio bands.
 
 /s/ Steve, N2CKH/AAR2EY
 
 A.4.4 ALE operational rules.
 The ALE system shall incorporate the basic operational rules listed
 in table A-V. Some of these
 rules may not be applicable in certain applications. For example,
 always listening is not
 possible while transmitting with a transceiver or when using a common
 antenna with a separate
 transmitter and receiver.
 
 TABLE A-V. ALE operational rules.
 1) Independent ALE receive capability (in parallel with other modems
 and simular audio receivers) (critical).
 2) Always listening (for ALE signals) (critical).
 3) Always will respond (unless deliberately inhibited).
 4) Always scanning (if not otherwise in use).
 5) Will not interfere with active channel carrying detectable traffic
 in accordance with table A-I (unless this
 listen call function is overriden by the operator or other controller).
 6) 

[digitalradio] Re: ALE QRM is minimal

2006-08-28 Thread Dave Bernstein
AA6YQ comments below


--- In digitalradio@yahoogroups.com, Steve Hajducek [EMAIL PROTECTED] 
wrote:

snip

 I have reviewed enough of the military documentation to understand
 that they employ dedicated ALE transceivers capable of much faster
 scanning rates.

Really?  Please enlighten me, I was under the impression that the ALE 
scan rates of 1, 2 and 5 ch/sec was it at present and that the future 
goal as stated in MIL-STD-188-141B was 10 ch/sec. The PC-ALE software 
supports 1, 2 and 5 ch/sec with an HF transceiver that is cable of 
all selections.

In a previous post, you stated that the scan rate for amateur ALE 
was 2 channels per second. Current military equipment supports 5 
channels per second. That's 2.5 time faster, is it not?
 

   As a result, sounding duration is signficantly
 reduced,

Sorry, but you will have to explain to me how Sounding duration  
decreases with an increase in the Scan Rate.

My understand from www.hflink.org is that a station must sound for 
an interval long enough to allow all receivers to scan all channels; 
a shorter sounding interval would result in some receivers failing to 
hear the sounding. If all receivers scan the same number of channels 
at a faster rate, then sounding duration can be decreased. is this 
not correct?


   and channel capacity increases in proportion.

Well not exactly. A 2 ch/sec scan rate allows you to cover the same 
number of channels faster than a 1 ch/sec scan rate and thus increase 
the odds of hearing a Sounding or a Linking call sooner.

A 2 ch/sec scan rate allows you to use half the sounding interval 
required by a 1 ch/sec scan rate. This doubles the number of 
supportable simultaneous users, does it not?


Running 2 or 5 ch/sec will also permit the station to be part of more 
than one ALE network at the same time, not an issue per see with 
Amateur Radio, but if two networks had scan groups of 10 channels 
each, you could scan both with excellent results.

We're discussing amateur ALE here.

 
The number of channels you scan does have an effect on your 
Soundings, you sound longer when you have more channels in the mix. 
There are variable here as we are now at a stage were you have 3 
generations of ALE. The latest ALE technology supports GPS time 
synchronization of the Scanning/Sounding which in the future will 
radically reduce BER/SNR data transfer for LQA ranking when all 
user's can support it.

Yes, that's the sort of approach I was asking about when I asked 
if there were techniques for reducing contention among competing 
sounders. PC-ALE would have to be extended to support this, and as 
you say its only effective if everyone uses it. 

 
   But one ham
 with an amateur transceiver limited to a 2 channel-per-second scan
 rate would force all ALE participants to sound for 20 seconds, even
 if their equipment could scan more rapidly. Do I have this right?

The details are to be found in MIL-STD-188-141B Appendix A. 
Regardless of the scan rate, if the controller Sounds based on number 
of channels in the scan group its less than 1 second per channel. The 
minimum redundant sound length (Trs) is equal to the standard 
one-word address leading call; that is, Trs = Tlc min = 2 Ta min = 2 
Trw = 784 ms. Thus for 12 channels it would be about 9.4 seconds 
depending on your address length being sent. The address length is 
based on an ALE Word which is 3 ASCII characters, for Amateur Radio 
applications we would being using 2 ALE Words as there are no 3 
character callsigns, whereas in the Military and Government world 
there are 3 character ALE Self Addresses being used. So W1AW, N2CKH 
and WB2XYZ are all 2 ALE Words were automatic padding is used to fill 
the second word. The least number of ALE Words the more efficient and 
reliable is the system. One would now want to use WB2XYZ/W6 to 
indicate they are in California. For AQC-ALE where many things were 
changed to make things even more efficient, a 2 ALE Word is the 
maximum allowed, whereas the original ALE allowed a 5 ALE Word (15 
character) Address to support the Military Automatic Digital Network 
(AUTODIN) system to directly link a Phone Patch call. Feel free to  
double check me with the standards, I am no expert on all this stuff 
and I am not perfect either, I make calculation errors often.

I don't see how the sounding duration can be independent of 
scanning rate (as discussed above), but using your 9.4 second 
sounding duration for 12 channels with each station sounding once per 
hour, (60*60)/9.4 yields a capacity of 383 users with 100% pilot 
channel utilization. Without synchronization or some other form of 
collision avoidance, the realistic number of simultaneous users would 
be in the range of 64 to 128.

This is considerably less than Bonnie's claim of 1000 simultaneous 
users - so either I'm misunderstanding something, or the claim is 
false.

   73,

   Dave, AA6YQ








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[digitalradio] Re: New to Digital HF -- PACTOR setup and hardware maybe needed???

2006-08-28 Thread Dave Bernstein
Agreed, there's no problem if you can own the OS; but on an end-
user's Windows PC, you can't do that.

  73,

  Dave, AA6YQ


--- In digitalradio@yahoogroups.com, Steve Hajducek [EMAIL PROTECTED] 
wrote:

 
 Hi Dave,
 
 I mentioned AMTOR as its timing is more robust that PACTOR I.
 
 As I have stated to the MARS-ALE users, the future version of that 
 tool when PACTOR I support is added ont he PCSDM will pretty much 
own 
 the OS, not a problem for our purposes as that one program running 
is 
 our only focus. GTOR is less demanding, but may be require the same 
 approach as PACTOR I. This is not required of DBM ARQ which is much 
 less demanding that GTOR.
 
 /s/ Steve, N2CKH
 
 At 06:43 AM 8/28/2006, you wrote:
 Precise timing isn't the issue, Steve. WinWarbler originally used
 GetTickCount() and QueryPerformanceCounter() in its CW generation
 code, but a high-resolution timer using the multimedia library is
 sufficiently accurate and more convenient. The problem is thread
 scheduling. WinWarbler uses SetPriorityClass to establish
 REALTIME_PRIORITY_CLASS and SetThreadPriority to establish
 THREAD_PRIORITY_TIME_CRITICAL . Nonetheless, there are many CPU-
 consumptive events that Windows insists on handling at higher
 priority -- such as displaying and moving the Task Manager window.
 Spin locks are not an acceptable solution IMHO; the user would find
 it disconcerting if the mouse pointer stopped following mouse
 movements during CW generation or Pactor-3 operation, for example.
 Yes, one could dedicate a headless Windows PC to executing one
 application like Pactor-3, but what would be the point? Even the 
most
 expensive Pactor-3 TNC would be cheaper and more compact to boot.
 
 I didn't comment on the feasibility of implementing AMTOR on 
Windows;
 Pactor-2 and Pactor-3 were the protocols I mentioned.
 
  From my days as a CPU architect and designer, I have lots of
 experience with real-time operating systems and applications -- but
 WinWarbler was my first encounter with extracting real-time
 performance from Windows. Any suggestions or pointers you have in
 this area would be appreciated.
 
  73,
 
   Dvae, AA6YQ








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[digitalradio] Re: New to Digital HF -- PACTOR setup and hardware maybe needed???

2006-08-28 Thread Dave Bernstein
Chas, the term modem is a contraction of modulator 
and demodulator; it purpose is the bidirectional conversion of 
digital signals to analog signals. There are many different kinds of 
modems, employing different modulation techniques to achieve 
different speeds and error rates over different transmission media. A 
modem used for internet dialup access, for example, is optimized for 
use with telephone lines and uses industry-standard signalling 
conventions so it can interoperate with any other telephone modem. 
The RTTY modem in in a TNC (Terminal Node Controller) uses the baudot 
code and RTTY mark and space tones and should be optimized for 
transmission and reception over HF. Circuits that modulate and 
demodulate PSK, FSK, Pactor, Amtor, or Olivia are all referred to as 
modems, but their implementations are radically different. Multi-mode 
TNCs typically employ a microprocessor and perhaps a programmable DSP 
circuit; software running on this one set of hardware components can 
thus implement multiple protocols -- but only one protocol is 
typically running at any one time.

The modem in your PC is most likely a telephone modem; it may also 
have facsimile capabilities.

The Icom 756 definitely does not include a modem; I doubt that the 
736 or 746 do either. More recent Icom transceivers like the 7800 and 
756 Pro 3 include a dedicated RTTY modem; to my knowledge, this modem 
cannot be used to encode or decode any other protocol.

You mentioned the Icom CI-V bus in an earlier post. This is the means 
by which an application running on a PC can control an Icom 
transceiver -- read or write its frequency or mode, etc. It is not a 
modem -- its a simple serial protocol using open-collector TTL levels 
(a binary 0 is represented by 0 volts, and a binary 1 is represented 
by 5 volts). To use this with a PC serial port, a level converter to 
RS-232 levels is required.

   73,

   Dave, AA6YQ

--- In digitalradio@yahoogroups.com, [EMAIL PROTECTED] wrote:

 On Mon, 28 Aug 2006 00:17:06 -0700, Chris Jewell
 [EMAIL PROTECTED] wrote:
 Suppose you're using your sound card as a modem to receive Pactor I
 data.  Your sound card takes care of turning tones from the 
receiver
 into 1s and 0s.  There's no problem there.
 
 actually, there is a modem in the 736.
 why can't that handle the packet checking tasks?
 it IS an FSK system but should be available, using HRD to run it, 
to handle
 any modem duties in the txcvr.   Also, I THINK, that there is a 
high speed
 modem built into my mobo as well as 10/100 etherlink.
 
 I am just trying to figure out why I have to add another piece of 
hardware to
 the system.  we KNOW that the latest, deep pocket ICOMs like the 
756P3 are
 fully set up with an internal TNC, or so I have been assured by 
some of
 those elmers/gurus (the ones who do not agree with each other G)
 The reason I am here is to try to find out what is true and what is 
bushwa and
 to simplify the shack to as great an extent as possible.  My 
experience is
 that the more hardware you add, the greater the load on output of 
final
 product.  that applies to anything with a pretty limited source of 
power, be
 it computing, motive or whatever.
 
 I truly do appreciate all the information I am getting here.
 
 Again, do I HAVE TO HAVE a TNC with an IC736, 746 or 756?
 
 thanks
 73/chas
 --
 K5DAM  Houston  EL29fuAAR6TU
 http://tinyurl.com/df55x (BPL Presentation)








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Re: [digitalradio] New to Digital HF -- PACTOR setup and hardware maybe needed???

2006-08-28 Thread Chris Jewell
KV9U writes:
  Chris,
  
  What is your view on using pipelined programming such as what was used 
  in the SCAMP mode to get around this issue with moving the ACK to the 
  next packet. The main penalty is latency for the user, but it seems 
  manageable.

I haven't read any detailed specs of the SCAMP protocol, only vague
descriptions, and I understand that the source code is not available
for public inspection at this time, but in principle a pipelined
approach seems like a good way to handle the problem of OS latency in
ACK/NAK at the application level, without requiring expert hackery in
low-level OS details.  My previous reply was just an attempt to
describe the problem of non-realtime operating systems and lockstep
ARQ modes such as Pactor for someone who had asked.

I note that another reply said that there is a alternative
real-time-capable kernel for Linux systems, so for those willing and
able to use Linux, the ARQ problem may be already solved.  However, a
solution available to the Windows and Mac users seems like a good
thing even so.

In TCP, the receiving host periodically sends an ACK that responds NOT
to a specific incoming packet, but instead says I have received
correctly everything up to byte number N.  If the sending host
doesn't get an ACK up through the bytes in a given packet within a
certain interval, it resends the data and all following data.  This
allows communication to succeed eventually if some ACKs or NAKs don't
get back to the sending station.

Of course, TCP/IP was conceived for full-duplex media: the sender can
keep the outgoing pipe full, provided the ACKs arrive soon enough.
The window size (how many bytes ahead of the ACK the sender may send)
needs to be larger for faster circuit, and also larger for a circuit
with a long round-trip time, such as via geosynch satellites or with
many overloaded routers in the path; the window can be smaller for
terrestrial circuits with few routers between the communicating hosts.
Partly because incorrect packets are going to be more frequent on HF
radio than on land lines, we probably should not duplicate the TCP
solution in ham radio, but it gives us a starting point to think about.

A system for half-duplex HF radio circuits has to provide for periodic
turnaround.  Short packets with frequent ACK/NAK cycles a la Pactor or
GTOR allow the receiver to promptly notify the sender of changing
propagation on the channel.  A deep pipeline postpones that feedback,
so more data may need to be resent when the circuit is unstable, BUT
also makes it straightforward to implement on a typical multitasking
OS.  It's all a matter of trade-offs.

Given that we deal with noiser circuits than were envisioned by the
designers of TCP/IP, we probably want a fixed (or periodically
adjusted) packet size, and a fixed number of packets sent before
turnaround.  The ACK would then say which of the packets arrived clean
and which need to be resent.  The sender could then resend those
packets which need it, and add some more new ones to make up the
agreed number to be placed in the pipeline.  (For the sake of
argument, how about 8 Pactor-sized packets followed by an ACK window 8
times as long as Pactor uses?)

This is off the top of my head: if anyone has already been thinking
more deeply about this and has better suggestions, by all means offer
them.  I'm an old computer geek but a new ham: I'm happy to learn
about either computers or radio from anyone who can improve my
understanding.

-- 
73 DE AE6VW Chris Jewell[EMAIL PROTECTED]Gualala CA USA 95445


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[digitalradio] -tor modes and PCs

2006-08-28 Thread jhaynesatalumni
I'm willing to believe that the timing tolerances in -tor modes
are so tight that ordinary PC operating systems cannot cope with
them the way a dedicated processor can.  What I don't understand 
is why the tolerances need to be so tight.  The transmitter sends
a packet and then listens for an ACK or NAK.  Why can't it wait
arbitrarily long?

There are protocols for wire transmission e.g. IBMs Bi-Sync
which worked in the days of modems that could only transmit in
one direction at a time.  These used old slow computers to
run the protocol.  





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[digitalradio] -tor modes and PCs

2006-08-28 Thread Chris Jewell
jhaynesatalumni writes:
  I'm willing to believe that the timing tolerances in -tor modes
  are so tight that ordinary PC operating systems cannot cope with
  them the way a dedicated processor can.  What I don't understand 
  is why the tolerances need to be so tight.  The transmitter sends
  a packet and then listens for an ACK or NAK.  Why can't it wait
  arbitrarily long?

The ACK time could be made as long as you like, but the throughput
would suffer accordingly.

For example, with Pactor I, (according to p. 9-24 of the 2005 ARRL
Handbook), the sender sends the packet in 0.96 seconds, then
propagation delays and receipt of the ACK takes 0.29 seconds, for a
total of 1.25 seconds per packet.  If we increase the ACK delay to be
the same as the transmit time, the total time per packet would be 1.92
seconds for the same amount of data as Pactor I sends in 1.25 second,
and the throughput would be 1.25/1.96 or approximately 0.65 times what
the present protocol delivers.

Is it doable?  Yes.  Would most hams want it?  I have my doubts.

To get the same throughput with a longer ACK time, you have to make
the transmit time longer too, so it bears the same relationship to the
total time as it does now.  That means either a much longer data
packet, or a pipelined group of packets covered by a single ACK.  The
longer the packet, the greater chance that a static crash or other
event will corrupt the packet, so we're back to talking about
pipelined packets.

-- 
73 DE AE6VW Chris JewellGualala CA USA



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Re: [digitalradio] Re: Timing requirements of digital ARQ

2006-08-28 Thread Patrick Lindecker
Hello Rick,

An additive remark to Dave. In Multispk, I have added Pactor 1 (RX/TX) but only 
in FEC. I would have liked to propose an ARQ Pactor1 but it is impossible. A 
bit in Pactor1 lasts 10 ms or 5 ms (200 bauds). This obliges to have a 
precision of at least  2 ms, which would be easy under DOS but impossible under 
Windows as you don't have a sample after sample input as under DOS, but 
buffers of datas. You can reduce the buffers length, perhaps down to an 
equivalent of 50 ms, on powerful PC but not less...so FEC (i.e asynchronous 
exchange of datas as in Pactor 1 FEC or in Pax) is the only solution. 

73
Patrick




  - Original Message - 
  From: Dave Bernstein 
  To: digitalradio@yahoogroups.com 
  Sent: Monday, August 28, 2006 6:35 PM
  Subject: [digitalradio] Re: Timing requirements of digital ARQ


  To use your example, Rick, if Windows introduces a 10 ms delay from 
  the time when you strike a button to initiate transmission in 
  MultiPSK until the point where Commander sends a CI-V Transmit! 
  command to your 756 Pro2, you'd never notice. However, such a delay 
  in confirming reception of a Pactor-3 packet would break the protocol.

  The SCS PTC-II Pro, which supports both Pactor-2 and Pactor-3, 
  utilizes a Motorola (now Freescale) 68360 clocked at 25 MHz and a 
  56303 24-bit DSP clocked at 100 MHz; see

  http://www.globalmarinenet.net/ptc.htm

  The 68360 is a descendant of the venerable 68000; its probably being 
  used for administrative tasks like running the serial port and 
  command line interface. The 56303 DSP does the heavy lifting. While 
  its DSP instruction set is advantageous, its performance is modest by 
  current standards; see

  http://www.bdti.com/articles/benchmark_icspat99.pdf#search=%22dsp%
  20benchmarks%22

  I am unable to locate a benchmark comparing the performance of 
  dedicated DSP chips like the 56303 DSP with general purpose 
  microprocessors, but I suspect that a 1 GHz Pentium could handle the 
  processing load.

  73,

  Dave, AA6YQ

  --- In digitalradio@yahoogroups.com, KV9U [EMAIL PROTECTED] wrote:
  
   Since I am not a programmer, other than taking some rudimentary 
  courses, 
   reading some of Don Lancaster's books, and knowing that it is not 
   something I could ever do very well, something still doesn't seem 
  right 
   to me when it comes to the claim that computers just can not meet 
  the 
   timing requirements of ARQ digital modes.
   
   When I use the computer to key my ICOM 756 Pro 2 with Dave's DX Lab 
   Commander software in order to run other programs that interface 
  with 
   Commander (such as Multipsk which is my main digital sound card 
   program), it does not seem to have much latency at all. And if one 
  could 
   key relatively slow CW as is done with the software for the 
  SDR1000, why 
   would that not be almost a magnitude faster than you would need for 
   adequate ARQ switching speed?
   
   How many ms do you need?
   
   Rick, KN6KB, the inventor of SCAMP, has said that the power in the 
   typical Windows OS computer we use is something like a magnitude 
  less 
   than that available in the dedicated SCS modems and that is why 
  they 
   perform so well compared to a computer (for the dedicated part). I 
   wonder where the dividing line will come so that computers will at 
  least 
   match the SCS type units?
   
   73,
   
   Rick, KV9U
   
   
   Dave Bernstein wrote:
   
   The issue is control over the operating system's scheduling 
   decisions. There are real-time versions of Linux that are 
  comparable 
   in this dimension to the firmware running in a TNC; given 
  sufficient 
   CPU horsepower, a Pactor-2 or Pactor-3 implementation on realtime 
   Linux is feasible.
   
   The Windows scheduler cannot be adequately controlled for use 
  timing-
   critical applications; as a result, the response time requirements 
  of 
   protocols like Pactor-2 and Pactor-3 cannot be guaranteed, no 
  matter 
   how fast the CPU. Generating CW with consistent timing is a 
  challenge 
   on this platform.
   
73,
   
Dave, AA6YQ
   

   
  



   

[Non-text portions of this message have been removed]



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[digitalradio] Re: -tor modes and PCs

2006-08-28 Thread Dave Bernstein
In the case of Pactor-2 and Pactor-3, the developers knew they were 
running on dedicated processors with complete control over scheduling, 
so there was no reason to reduce performance by unnecessarily extending 
turnaround time or pipelining control messages (which extends recovery 
when an error occurs). If you're in the business of selling standalone 
TNCs, as these developers are, then you exploit every strength afforded 
by this approach. The last thing you'd do is design a protocol that 
could be implemented without your hardware!

IBM Bisync? I know it all too well. The first commercial product I 
designed was a hardware interface for this protocol. Dating myself, it 
was implemented with TTL MSI -- before the availability of large scale 
integration or customizable logic. 

   73,

   Dave, AA6YQ







--- In digitalradio@yahoogroups.com, jhaynesatalumni [EMAIL PROTECTED] 
wrote:

 I'm willing to believe that the timing tolerances in -tor modes
 are so tight that ordinary PC operating systems cannot cope with
 them the way a dedicated processor can.  What I don't understand 
 is why the tolerances need to be so tight.  The transmitter sends
 a packet and then listens for an ACK or NAK.  Why can't it wait
 arbitrarily long?
 
 There are protocols for wire transmission e.g. IBMs Bi-Sync
 which worked in the days of modems that could only transmit in
 one direction at a time.  These used old slow computers to
 run the protocol.








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RE: [digitalradio] New to Digital HF -- PACTOR setup and hardware maybe needed???

2006-08-28 Thread DuBose Walt Civ AETC CONS/LGCA
Back in the late 80's when Mil-STD-188-110 was still in development, a Harris 
RF Comm Gp engineer/programmer recommended that you send a frame with X number 
of numbered packets with a CRC.  The receiving station would only NAK the 
packets it didn't receive.  The transmitting stations would start transmitting 
another frame with the missed packets first and then the new packets.  The 
total of the packets, missed and new, would be less than the original number of 
packets.  When no NAKs were received, the transmitting station would start 
sending more packets in a frame.

Walt/K5YFW

-Original Message-
From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED]
Sent: Monday, August 28, 2006 2:28 PM
To: digitalradio@yahoogroups.com
Subject: Re: [digitalradio] New to Digital HF -- PACTOR setup and
hardware maybe needed???


KV9U writes:
  Chris,
  
  What is your view on using pipelined programming such as what was used 
  in the SCAMP mode to get around this issue with moving the ACK to the 
  next packet. The main penalty is latency for the user, but it seems 
  manageable.

I haven't read any detailed specs of the SCAMP protocol, only vague
descriptions, and I understand that the source code is not available
for public inspection at this time, but in principle a pipelined
approach seems like a good way to handle the problem of OS latency in
ACK/NAK at the application level, without requiring expert hackery in
low-level OS details.  My previous reply was just an attempt to
describe the problem of non-realtime operating systems and lockstep
ARQ modes such as Pactor for someone who had asked.

I note that another reply said that there is a alternative
real-time-capable kernel for Linux systems, so for those willing and
able to use Linux, the ARQ problem may be already solved.  However, a
solution available to the Windows and Mac users seems like a good
thing even so.

In TCP, the receiving host periodically sends an ACK that responds NOT
to a specific incoming packet, but instead says I have received
correctly everything up to byte number N.  If the sending host
doesn't get an ACK up through the bytes in a given packet within a
certain interval, it resends the data and all following data.  This
allows communication to succeed eventually if some ACKs or NAKs don't
get back to the sending station.

Of course, TCP/IP was conceived for full-duplex media: the sender can
keep the outgoing pipe full, provided the ACKs arrive soon enough.
The window size (how many bytes ahead of the ACK the sender may send)
needs to be larger for faster circuit, and also larger for a circuit
with a long round-trip time, such as via geosynch satellites or with
many overloaded routers in the path; the window can be smaller for
terrestrial circuits with few routers between the communicating hosts.
Partly because incorrect packets are going to be more frequent on HF
radio than on land lines, we probably should not duplicate the TCP
solution in ham radio, but it gives us a starting point to think about.

A system for half-duplex HF radio circuits has to provide for periodic
turnaround.  Short packets with frequent ACK/NAK cycles a la Pactor or
GTOR allow the receiver to promptly notify the sender of changing
propagation on the channel.  A deep pipeline postpones that feedback,
so more data may need to be resent when the circuit is unstable, BUT
also makes it straightforward to implement on a typical multitasking
OS.  It's all a matter of trade-offs.

Given that we deal with noiser circuits than were envisioned by the
designers of TCP/IP, we probably want a fixed (or periodically
adjusted) packet size, and a fixed number of packets sent before
turnaround.  The ACK would then say which of the packets arrived clean
and which need to be resent.  The sender could then resend those
packets which need it, and add some more new ones to make up the
agreed number to be placed in the pipeline.  (For the sake of
argument, how about 8 Pactor-sized packets followed by an ACK window 8
times as long as Pactor uses?)

This is off the top of my head: if anyone has already been thinking
more deeply about this and has better suggestions, by all means offer
them.  I'm an old computer geek but a new ham: I'm happy to learn
about either computers or radio from anyone who can improve my
understanding.

-- 
73 DE AE6VW Chris Jewell[EMAIL PROTECTED]Gualala CA USA 95445


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Re: [digitalradio] -tor modes and PCs

2006-08-28 Thread Steve Hajducek
At 11:41 AM 8/28/2006, you wrote:
I'm willing to believe that the timing tolerances in -tor modes
are so tight that ordinary PC operating systems cannot cope with
them the way a dedicated processor can.  What I don't understand
is why the tolerances need to be so tight.  The transmitter sends
a packet and then listens for an ACK or NAK.  Why can't it wait
arbitrarily long?

There are protocols for wire transmission e.g. IBMs Bi-Sync
which worked in the days of modems that could only transmit in
one direction at a time.  These used old slow computers to
run the protocol.

That is pretty much the way it is with the ALE DTM ARQ, DBM ARQ 
protocols and most of the ARQ protocols on the high speed 
MIL-STD-188-110x PSK serial tone modem.

The FSK one's will run on just about any CPU and PC Sound Device when 
an 8Khz sample clock is used.

The '188-110 modem is a CPU/Memory hog and requires a 9.6Khz sample 
clock, to have both the FSK/PSK modems active at the same time on the 
same PC Sound Device you need a common ground so 48Khz is used as its 
dividable by both 8k and 9.6k. We have a version of MARS-ALE called 
Legacy Edition that only has the FSK modem at 8Khz and then both 
modem at 48Khz, thus the MARS-ALE LE version runs on a 386 and 
Windows 98SE and that DBM ARQ is a real performer at a raw 125 baud 
and deep interleaving.

/s/ Steve, N2CKH




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Re: [digitalradio] -tor modes and PCs

2006-08-28 Thread Steve Hajducek

Hi Chris,

Take a look at DBM ARQ in 
http://www.n2ckh.com/MARS_ALE_FORUM/MIL-STD-188-141B.pdf starting on 
page 178, it really lends itself to the PCSDM and its works 
fantastic, I love GTOR, more so than PACTOR I since the day I bought 
my first KAM Plus (I like my KAM XL a bit better though) and DBM ARQ 
( GTOR is a kissing cousin) is every bit as good and in some ways 
better. Here's a comparison of the TOR's and DMB ARQ timing:

ProtocolCycle LengthData Length Ack Length   NOTE: 
All in seconds:

AMTOR   0.450.210.07

PACTOR  1.250.960.12

GTOR2.401.920.16

DBM ARQ variablevariable1.57

If for no other reason, anyone interested in a robust FSK ARQ 
protocol (there DBM BRD which is just FEC that is great as well) 
should give it a try in any version of PC-ALE, no Scanning, no 
Sounding required, just you and a buddy establish a link and then 
send your message with DBM, pretty much like establishing a link with 
GTOR or PACTOR but no constant channel diddling while linked, its so 
much less annoying to others tuning by or having a QSO in the 
distance, the only transmitting is the Linking and then the sending 
of the data and the handshaking and the throughput is just as fast.

As my Mother used to say and was often right, Try it, you'll like it!

/s/ Steve, N2CKH

At 07:23 PM 8/28/2006, you wrote:

The ACK time could be made as long as you like, but the throughput
would suffer accordingly.

For example, with Pactor I, (according to p. 9-24 of the 2005 ARRL
Handbook), the sender sends the packet in 0.96 seconds, then
propagation delays and receipt of the ACK takes 0.29 seconds, for a
total of 1.25 seconds per packet.  If we increase the ACK delay to be
the same as the transmit time, the total time per packet would be 1.92
seconds for the same amount of data as Pactor I sends in 1.25 second,
and the throughput would be 1.25/1.96 or approximately 0.65 times what
the present protocol delivers.

Is it doable?  Yes.  Would most hams want it?  I have my doubts.

To get the same throughput with a longer ACK time, you have to make
the transmit time longer too, so it bears the same relationship to the
total time as it does now.  That means either a much longer data
packet, or a pipelined group of packets covered by a single ACK.  The
longer the packet, the greater chance that a static crash or other
event will corrupt the packet, so we're back to talking about
pipelined packets.

--
73 DE AE6VW Chris JewellGualala CA USA




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RE: [digitalradio] The digital throughput challenge on HF

2006-08-28 Thread John Champa
Why?  Because your digital voice QSO sounds like noise to SSBers?


From: John Becker [EMAIL PROTECTED]
Reply-To: digitalradio@yahoogroups.com
To: digitalradio@yahoogroups.com
Subject: RE: [digitalradio] The digital throughput challenge on HF
Date: Sun, 27 Aug 2006 20:29:05 -0500

You should try DIGITAL VOICE once.
There is 1000% chance that your QSO will be
QRM'ed

At 04:43 AM 8/25/2006, you wrote:

 The problem is QRM. Consisting of PACTOR, MFSK, OLIVIA, PSK31, and on 30
 meters also SSB signals coming on frequency during your qso.











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[digitalradio] What is ALE?

2006-08-28 Thread expeditionradio
An Article on the HFLINK.COM website explains ALE for amateur radio
operators. The PCALE software to run ALE on your ham transceiver is
also available for download on HFLINK.COM website. 

ALE on HF in the Amateur Radio Service
Selective Calling and Automatic Linking for Voice, Text, Image, CW,
and Data

What is ALE?
ALE is Automatic Link Establishment. With the capability to call up a
specific station, a group of stations, a net... or just CQ, ALE is a
versatile system for connecting operators on the air for voice, data,
text, instant messaging, CW, or image QSOs. A radio operator
initiating an ALE call on HF or VHF, can... 
read more click here:
http://hflink.com/articles

ALE in Ham Radio
For the past 5 years, a group of ham operators has joined together for
experiments and communications in Amateur Radio ALE and Selective
Calling. The number of hams has grown from just a handful active in
2001, to the hundreds of... 
read more click here:
http://hflink.com/articles

How ALE Works
Each ham radio ALE station uses the operator's callsign as an address
in the ALE controller.When not actively in a QSO with another station,
each HF SSB transceiver constantly scans through a list of
frequencies, listening for its callsign. To reach a specific station,
the caller simply enters the callsign just like... (read more) 

What is the Bandwidth of the ALE Signal?  The emission type and
bandwidth of the ALE 8FSK is ...
read more click here:
http://hflink.com/articles


73---Bonnie KQ6XA

.






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RE: [digitalradio] What is ALE?

2006-08-28 Thread rattray
Thank you Bonnie - 73 Bruce.

72/73 - Bruce ve5rc/ve5qrp - QRP-C#1, QRP-L#886, A1 Operator
Enter QRP-Canada's RUN with RAC contest -
    details - http://www.qrp-canada.com 
 



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