AllStarLink is a network of Amateur Radio repeaters, remote base stations and
hot spots accessible to each other via Voice over Internet Protocol.
AllStarLink:
https://www.allstarlink.org/
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The creator of AllStarLink is Jim Dixon, WB6NIL (Silent Key). Jim passed away on
December 16, 2016.
His objective was to design a radio interface that could interconnect
a computer and the typical interface signals of two-way radios. Jim designed and
built the first AllStar style radio interface hardware modules and developed a
software application based upon
Asterisk, a free and open source framework
for building communications applications. Jim developed the app_rpt module which allowed
the open source PBX system, Asterisk, to function as a repeater controller.
Jim developed a radio interface that could interconnect with a PC internally and
the typical interface signals of a two-way radio:
The story of App_Rpt development can be found in an article by Jim Dixon WB6NIL
himself on the AllStarLink History page:
https://wiki.allstarlink.org/wiki/History
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There are two providers of the AllStar System:
Both offerings are build on Asterisk.org, an
open source communications project that lets you create telephony apps for
IP PBXs, VoIP Gateways.
The most common build is on the Raspberry Pi, RPi3. The Allstarlink distribution is
built on the Raspberian OS and Hamvoip is built on Archlinux OS.
_ _ _____ _ _ _ _ /\ | | |/ ____| | | | (_) | | / \ | | | (___ | |_ __ _ _ __ | | _ _ __ | | __ / /\ \ | | |\___ \| __|/ _` | '__| | | | | '_ \| |/ / / ____ \| | |____) | |_| (_| | | | |____| | | | | < /_/ \_\_|_|_____/ \__|\__,_|_| |______|_|_| |_|_|\_\The AllStar Home Page: https://www.allstarlink.org/
https://www.qsl.net/w2ymm/allstar1.html
In order to become a member of the AllStar Link Network, you must first
Register to become user of the Portal system. To qualify for membership,
you must be a holder of a valid Amateur Radio license.
A URI or RIM (Radio Interface Module) is required.
I purchased the Repeater Builder RIM-Lite,
Repeater Builders's RB-RIM-Lite (local)
An interface can be constructed using a cheap USB sound FOB. Search the internet
for conversion instructions.
Wifi and DHCP are enabled by default and the RPi will attempt to obtain an IP lease.
Secure Shell was not enabled by default.
Try ssh repeater@192.168.5.213
the ssh port is the default port: 22
Login: repeater
Passsword: allstarlink # The is the default password
On first login you will be requested to reset the password for the "repeater" account.
The repeater account has sudo privileges (sudo -s)
There may have been updates to the operating system and AllStarLindvswk since this image was built.
This configures parameters for the USB radio interface. It is far better to
perform this configuration using the simpleusb-tune-menu
The Connection Map is useful to confirm that connections have been made between node.
It also gives an understanding of the AllStar communication around the world at
any given time.
When connected check the AllStar Connection Map:
http://allstarmap.org/allstarmap.html
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Because we use the ADPCM (Adaptive Differential Pulse-Code Modulation) g726aal2
codec with Asterisk, each connection from/to
a remote node will require about 55 kilobits/sec (at the IP packet level) of
bandwidth in both directions. The frequency response when ADPCM is used will
be very close to telephone voice grade (telephone voice grade is defined as
3db points at 300-3400Hz, 1000Hz 0db reference).
Due to the fact that the system is TDM (Time Domain Multiplexed) and there is
overhead involved to perform DSP functions in Asterisk, there is audio delay
for both local connections and there is additional audio delay from
connections made over the Internet.
Adaptive differential pulse-code modulation (ADPCM) is a variant of
differential pulse-code modulation (DPCM) that varies the size of the
quantization step, to allow further reduction of the required data bandwidth
for a given signal-to-noise ratio.
Typically, the adaptation to signal statistics in ADPCM consists simply of
an adaptive scale factor before quantizing the difference in the DPCM encoder.
ADPCM was developed in the early 1970s at Bell Labs for voice coding,
by P. Cummiskey, N. S. Jayant and James L. Flanagan.
Source:https://en.wikipedia.org/wiki/Adaptive_differential_pulse-code_modulation
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Active simpleusb stanza: [usb] USB device String: 1-1.1.3:1.0 Card is: 0 Name is: usb Rx Level: 500 Rx no-delay: 0 Rx audio-delay: 0 Tx A Level: 700 Tx B Level: 700 preemphasis: no deemphasis: no plfilter: no dcsfilter: no rxboost: no PTT: Active LOW PTT status: Un-Keyed CTCSS (input): Ignored COS (input): CM108/CM119 Active LOW COS (test): Un-Keyed COS (composite): Un-Keyed
Incoming connections are made from any Echolink node in the usual manner. An app_rpt user dials an additional prefix digit in the connect command to distinguish between Allstar (2), Echolink (3) node numbers.
I changed my duplex setting in the rpt.conf file and now that I'm transmitting
telemetry and courtesy tones I am back on the map. Switching off the telemetry
is how I fixed the chasing the tail problem. The docs say that it shouldn't
matter if I dont advertise my node people should still be able to connect but
now no one can connect and I am not on the map at
http://allstarmap.org/allstarmap.html
To check you have no registration issues you do a check_reg.sh from the bash prompt.
To check you have a route to the node you are wishing to contact you do "dns-query <node>"
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This is a write up by my good friend. Rick and I have colaborated on many
projects and he has always challanged me on to greater heights.
AllStar–Analogue-FM-RoIP-Rick-Perks-VK4HC-S79RP.pdf (local)
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Because we use the ADPCM (g726aal2) codec in Asterisk, each connection from/to a
remote node will require about 55 kilobits/sec (at the IP packet level) of
bandwidth in both directions. Optionally, the GSM codec can be used and the
bandwidth requirement will be 35 kilobits/sec, but the audio quality will suffer.
The frequency response when ADPCM is used will be very close to telephone voice
grade (telephone voice grade is defined as 3db points at 300-3400Hz, 1000Hz
0db reference). Additionally, we optionally support the G.711 codec which is
superior ro ADPCM at the expense of bandwidth.
Source:
https://wiki.allstarlink.org/wiki/ASL_FAQ#How_much_bandwidth_is_required_per_connection.3F
The hamvoip code has the ILBC (Internet low bandwidth codec) installed. It
uses about 1/3 of the bandwidth of the G726AA codec which is most commonly
used for Allstar. Its audio quality is very good and very close to G726.
Most people would not be able to tell the difference. The reason we do not
use it by default is that it poorer "squelching" than G726. G726 on a poor
connection mutes dropouts whereas ILBC tends to allow the garbage sounds
through. I often recommend that cell phone hotspot users try ILBC but
anyone trying to conserve bandwidth can give it a try. Most would not have
that issue as voice is rather low bandwidth compared to most other things
we do on the Internet nowadays. Codec settings are in
/etc/asterisk/iax.conf. Ordering sets priority but if the called node does
not have the codec you request then it will take the next one down the list
until it is satisfied. I would recommend you try the ILBC codec and see if
it satisfies your needs.
Source:
http://lists.hamvoip.org/pipermail/arm-allstar/2017-May/005191.html
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