OP25j

 = 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 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.

[[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 per 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

 == Build asterisk with app_rpt ==

First obtain and unpack the sources

 - http://ohnosec.org/drupal/node/6   [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

http://www.voip-info.org/wiki/view/Asterisk+quickstart

 == 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 extension "1" in context {{{radio}}} (also in {{{/etc/asterisk/extensions.conf}}})

{{{

[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)

 = 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 usrp_tx.py receives data from chan_usrp.c [port 34001]

Example:

{{{

usrp_tx.py -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, {{{usrp_tx.py}}} 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:

{{{

soundcard_tx.py -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}}})