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max, 04/22/2017 04:04 PM


= SUMMARY = * Simultaneous multi-channel transmission/reception * Uses USRP and/or conventional receiver/transmitter hardware * Generic interface to Asterisk via app_rpt (full VOIP, radio control, and standard repeater functions) * Features: P25, analog NBFM with CTCSS * Expected to be fully "ROIP" [Radio Over IP] compatible

= Multi-channel reception using the USRP =

Assume we want to simultaneously receive the four signals shown below;
one conventional (analog) FM voice channel plus three P25
signals. The two P25 voice channels are to be IMBE-decoded whereas the
P25 data channel is to be sent to Wireshark after decoding. For all three voice
channels, we want to forward the received PTT and audio info to Asterisk app_rpt.
The received PTT [Push To Talk, or "keyed"] signal is a bit that ideally tracks
the state of the PTT key at transmitter, indicating the presence or absence of the
received signal.

Image(sa.png)

Fig. 1 - Spectrum of repeater input band (sample; diagram not to scale)

For now, it's necessary to edit the source code file manually to specify
the list of channels/modes to be received (file usrp_rx.py):

{{{
channels = [ {'freq':435.200e6, 'mode':'c4fm', 'port':32001}, {'freq':435.350e6, 'mode':'fm', 'port':32002, 'ctcss':97.4}, {'freq':435.600e6, 'mode':'cqpsk', 'port':23456, 'wireshark':1}, {'freq':435.775e6, 'mode':'cqpsk', 'port':32003}
]
}}}

Individual channels are defined one per line; note that all definition lines
except the last must end with a comma.

We choose a frequency somewhere close to the center of this band.
which will set the USRP's nominal receive frequency; this must also
be manually set in the source file:

{{{
center_freq = 435.500e6
}}}

Before running the receiver app, we * measure the current calibration error value (I use kalibrate) * determine the optimum USRP receiver gain value
The values used in this example are +1234 and 35, respectively.

We're now ready to start the receiver:

{{{
usrp_rx.py -RA -c 1234 -H 127.0.0.1 -g 35 -d 25
}}}

The receiver continuously monitors all four channels. For each of the three
voice channels, the audio and the PTT info ("key up" and "key down" events)
are forwarded to asterisk app_rpt over separate UDP channels.

= P25 and/or analog NBFM Reception using a discriminator-tapped receiver =

One or more disc-tapped conventional receivers may be used at the same time, and can
coexist with one or more USRP's.

An audio cable is connected between the disc-tap point in the receiver and the PC soundcard.

Single-channel reception is possible using disctap_rx.py.
The app dynmically auto-detects the modulation type (P25 or analog NBFM).

Audio and PTT events are forwarded to asterisk app_rpt over two separate UDP
channels, depending on modulation type: * received P25 audio is sent to asterisk on UDP port 32004 * when analog NBFM is received with the proper CTCSS tone (97.4 Hz), port 32005 is used

{{{
disctap_rx.py -i -A 0.05 -c 97.4 -H 127.0.0.1 -p 32004 -g 35 -d 25
}}}

The -g (gain) parameter is used to set the proper audio gain level. See
the hardware page for further guidance - this value is important for achieving
correct operation.

= Asterisk and app_rpt =

For all voice modes (IMBE and analog FM) the audio is transmitted
as frames over the UDP channel to and from Asterisk in the standard native audio format: * 50 frames per second * 160 audio samples per frame * 8000 samples / second * signed * 16-bit * linear

Installation

First, obtain and unpack the app_rpt source tree.

Second, locate the subdirectory named {{{asterisk/channels}}} in the source tree you just unpacked,
and copy the files {{{chan_usrp.c}}} and {{{chan_usrp.h}}} (from {{{src/lib}}}) to this subdirectory.

Next, build and install asterisk and app_rpt, and verify that chan_usrp is included

Configuration

We define five repeater nodes in {{{/etc/asterisk/rpt.conf}}}

{{{
[000]
rxchannel = usrp/127.0.0.1:34001:32001
duplex = 2
scheduler=scheduler
functions = functions-repeater
hangtime=0
authlevel = 0

[001]
rxchannel = usrp/127.0.0.1:34002:32002
duplex = 2
scheduler=scheduler
functions = functions-repeater
hangtime=0
authlevel = 0

[002]
rxchannel = usrp/127.0.0.1:34003:32003
duplex = 2
scheduler=scheduler
functions = functions-repeater
hangtime=0
authlevel = 0

[003]
rxchannel = usrp/127.0.0.1:34004:32004
duplex = 2
scheduler=scheduler
functions = functions-repeater
hangtime=0
authlevel = 0

[004]
rxchannel = usrp/127.0.0.1:34005:32005
duplex = 2
scheduler=scheduler
functions = functions-repeater
hangtime=0
authlevel = 0
}}}

Continuing the example of five voice channels from above, we define five repeater nodes (channels). Voice and PTT traffic that
is output by asterisk/app_rpt for RF transmission is forwarded to usrp_tx.py (see below) using UDP ports in the 3400x range.
Voice data received in usrp_rx.py and/or disctap_rx.py is forwarded to asterisk/app_rpt (chan_usrp.c) via ports in the 3200x range.

The driver invocation in {{{rpt.conf}}} is {{{
usrp/HISIP:HISPORT[:MYPORT]
HISIP is the IP address (or FQDN) of the GR app
HISPORT is the UDP socket of the GR app
MYPORT (optional) is the UDP socket that Asterisk listens on
for this channel
}}}

TIP: You can use the {{{usrp show}}} command to display status information from within the Asterisk CLI.

TIP: Another handy command is {{{rpt playback}}} to start transmission on a channel.

Channel Bank Configuration

Typically the audio links are terminated on channel banks which provide a standard interface to user
equipment. Commonly, this equipment places an "offhook" indication on the signalling circuit when it
wishes to initiate a radio transmission, and signals the end of the transmission by placing the circuit
in the "onhook" state. Standard audio transmission levels are defined at the channel bank interface.

A standard FXS port on the channel bank is defined in {{{/etc/asterisk/zapata.conf}}} with {{{
signalling=fxo_ls
immediate=yes
context = chan1
channel => 1
}}}

An offhook (PTT) signal from user equipment on the FXS channel bank port
(due to the {{{immmediate=yes}}}) starts processing in {{{/etc/asterisk/extensions.conf}}}:

{{{
[chan1]
exten => s,1,Dial(local/1@radio/n)
This jumps to exten "1" in
[radio]
exten => 1,1,rpt(000|D)
exten => 2,1,rpt(001|D)
exten => 3,1,rpt(002|D)
exten => 4,1,rpt(003|D)
exten => 5,1,rpt(004|D)
}}}

So the call resulting from the offhook (PTT) signal is routed to
extension "1" in context {{{[radio]}}} where it's connected to the desired
repeater node (channel). If the GR app is running it will initiate
radio transmission. An onhook signal on the FXS channel bank port
causes the end of the transmission by ending the asterisk call in
progress*. The {{{hangtime=0}}} setting in {{{rpt.conf}}} was used to reduce the
tail delay in this setup.

*The end of the transmission may be modified however, for example when
app_rpt appends an "ID" or if a "timeout" occurs.

Note: "Dumb" mode is used in these examples (theory: if it can't be made to work in its dumb mode, there's no prayer of getting smart mode to work)

= Transmission (USRP and conventional) =

TBC

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Updated by max about 7 years ago · 5 revisions

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