• 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 receive simultaneously 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.

Spectrum plot

Spectrum plot as received by a USRP:-

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

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: -RA -c 1234 -H -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
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 -i -A 0.05 -c 97.4 -H -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 per second * signed * 16-bit * linear

Build asterisk with app_rpt

As a prereq make sure the Linux kernel headers are installed, as zaptel installs kernel modules.

First obtain and unpack the sources

-   [use svn checkout to grab the "Asterisk Sources"]

Locate the toplevel directory - this should have several subdirectories including zaptel and asterisk .

Then locate the subdirectory named asterisk/channels ,
and copy the files chan_usrp.c and chan_usrp.h (from repeater/src/lib in the op25 sources) to this subdirectory.

Now, build and install asterisk and app_rpt

cd to the toplevel directory

cd zaptel && ./configure && make && sudo make install

and then (again from the toplevel directory)

cd asterisk && ./configure && make && sudo make install

For a first time installation of asterisk there are also other steps such as config file installation.
Before setting up rpt.conf you must first set up asterisk itself - see for example


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

rxchannel = usrp/
duplex = 2
functions = functions-repeater
authlevel = 0

rxchannel = usrp/
duplex = 2
functions = functions-repeater
authlevel = 0

rxchannel = usrp/
duplex = 2
functions = functions-repeater
authlevel = 0

rxchannel = usrp/
duplex = 2
functions = functions-repeater
authlevel = 0

rxchannel = usrp/
duplex = 2
functions = functions-repeater
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 (see below) using UDP ports in the 3400x range.
Voice data received in and/or is forwarded to asterisk/app_rpt (chan_usrp.c) via ports in the 3200x range.

The driver invocation in rpt.conf is

    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

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:

exten => s,1,Dial(local/1@radio/n)

This jumps to extension "1" in context radio (also in /etc/asterisk/extensions.conf)
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)

USRP Transmission

The multi-channel USRP transmitter app currently has some limitations: * Transmit channel spacing is at arbitrary 25 KHz intervals * Analog NBFM mode is not yet supported
Before running the app you must first determine (example values shown in square brackets - YMMV) * the USRP TX daughterboard ID (A or B) [A] * carrier frequency of the first ("center") TX channel [435.125 MHz] * number of channels to be transmitted [five] * the first UDP port number over which receives data from chan_usrp.c [port 34001]
Example: -TA -e -f 435.125e6 -n 5 -p 34001

Here we also request a FFT display. With port 34001 as the starting UDP port number and
five channels, listens on ports 34001-34005 for TX traffic coming from chan_usrp.c, as
configured in /etc/asterisk/rpt.conf (see above).

Soundcard Transmission

The soundcard TX program outputs an analog waveform that is suitable for application to the
modulator stage of an FM transmitter. It listens on the UDP port for audio frames sent to it
by asterisk/chan_usrp. The analog audio thus received is encoded by the soundcard TX app to
IMBE voice code words, which are in turn assembled into P25 voice frames (LDU1/LDU2's). The
resulting 4800 baud symbol stream is then RRC filtered and shaped according to the P25 spec.
This signal is output to the sound card (and optionally to an on-screen oscilloscope using
the -e option).

This app may be run on the same server as asterisk or on a different machine.

Currently only a single channel is supported (per invocation of soundcard_tx). If more than
one TX channel is to be used simultaneously, run a another copy of the app (either on a different
host or using a different UDP port).

Syntax: -g 0.3 -p 32001

The gain value (0.3) would cause the output envelope (ranging from -3 to +3) to be scaled to
fit within the standard band from -1 to +1. Values larger than 0.3 may cause clipping and distortion.

The final output amplitude to the radio can also be adjusted in discrete steps using the audio mixer
application (e.g., alsamixer)

Updated by matt about 6 years ago · 11 revisions

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