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matt, 10/26/2017 12:54 AM

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h1. SUMMARY
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 * Simultaneous multi-channel transmission/reception
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 * Uses USRP and/or conventional receiver/transmitter hardware
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 * Generic interface to Asterisk via app_rpt (full VOIP, radio control, and standard repeater functions)
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 * Features: P25, analog NBFM with CTCSS
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 * Expected to be fully "ROIP" [Radio Over IP] compatible
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h2. Multi-channel reception using the USRP
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Assume we want to receive simultaneously the four signals shown below;
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one conventional (analog) FM voice channel plus three P25
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signals.  The two P25 voice channels are to be IMBE-decoded whereas the
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P25 data channel is to be sent to Wireshark after decoding.  For all three voice
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channels, we want to forward the received PTT and audio info to Asterisk app_rpt.
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The received PTT [Push To Talk, or "keyed"] signal is a bit that ideally tracks
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the state of the PTT key at transmitter, indicating the presence or absence of the
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received signal.
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!spectrum.png!
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Fig. 1 - Spectrum of repeater input band (sample; diagram not to scale)
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For now, it's necessary to edit the source code file manually to specify
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the list of channels/modes to be received (file usrp_rx.py):
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<pre>
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channels = [
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	{'freq':435.200e6, 'mode':'c4fm',  'port':32001},
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	{'freq':435.350e6, 'mode':'fm',    'port':32002, 'ctcss':97.4},
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	{'freq':435.600e6, 'mode':'cqpsk', 'port':23456, 'wireshark':1},
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	{'freq':435.775e6, 'mode':'cqpsk', 'port':32003}
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]
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</pre>
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Individual channels are defined one per line; note that all definition lines
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except the last must end with a comma.
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We choose a frequency somewhere close to the center of this band.
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which will set the USRP's nominal receive frequency; this must also
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be manually set in the source file:
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<pre>
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center_freq = 435.500e6
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</pre>
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Before running the receiver app, we
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 * measure the current calibration error value (I use kalibrate)
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 * determine the optimum USRP receiver gain value
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The values used in this example are +1234 and 35, respectively.
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We're now ready to start the receiver:
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<pre>
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usrp_rx.py -RA -c 1234 -H 127.0.0.1 -g 35 -d 25
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</pre>
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The receiver continuously monitors all four channels.  For each of the three
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voice channels, the audio and the PTT info ("key up" and "key down" events)
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are forwarded to asterisk app_rpt over separate UDP channels.
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 = P25 and/or analog NBFM Reception using a discriminator-tapped receiver =
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One or more disc-tapped conventional receivers may be used at the same time, and can
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coexist with one or more USRP's.
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An audio cable is connected between the disc-tap point in the receiver and the PC soundcard.
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Single-channel reception is possible using disctap_rx.py.
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The app dynmically auto-detects the modulation type (P25 or analog NBFM).
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Audio and PTT events are forwarded to asterisk app_rpt over two separate UDP
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channels, depending on modulation type:
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 * received P25 audio is sent to asterisk on UDP port 32004
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 * when analog NBFM is received with the proper CTCSS tone (97.4 Hz), port 32005 is used
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<pre>
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 disctap_rx.py -i -A 0.05 -c 97.4 -H 127.0.0.1 -p 32004 -g 35 -d 25
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</pre>
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The -g (gain) parameter is used to set the proper audio gain level.  See
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the hardware page for further guidance - this value is important for achieving
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correct operation.
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h2. Asterisk and app_rpt
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For all voice modes (IMBE and analog FM) the audio is transmitted
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as frames over the UDP channel to and from Asterisk in the standard native audio format:
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 * 50 frames per second
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 * 160 audio samples per frame
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 * 8000 samples per second
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 * signed
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 * 16-bit
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 * linear
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h3. Build asterisk with app_rpt
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As a prereq make sure the Linux kernel headers are installed, as zaptel installs kernel modules.
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First obtain and unpack the sources
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 - http://ohnosec.org/drupal/node/6   [use @svn checkout@ to grab the "Asterisk Sources"]
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Locate the toplevel directory - this should have several subdirectories including @zaptel@ and @asterisk@ .
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Then locate the subdirectory named @asterisk/channels@ ,
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and copy the files @chan_usrp.c@ and @chan_usrp.h@ (from @repeater/src/lib@ in the op25 sources) to this subdirectory.
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Now, build and install asterisk and app_rpt
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@cd@ to the toplevel directory
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@cd zaptel && ./configure && make && sudo make install@
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and then (again from the toplevel directory)
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@cd asterisk && ./configure && make && sudo make install@
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For a first time installation of asterisk there are also other steps such as config file installation.
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Before setting up @rpt.conf@ you must first set up asterisk itself - see for example
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http://www.voip-info.org/wiki/view/Asterisk+quickstart
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h2. Configuration
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We define five repeater nodes in @/etc/asterisk/rpt.conf@
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<pre>
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[000]
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rxchannel = usrp/127.0.0.1:34001:32001
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duplex = 2
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scheduler=scheduler
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functions = functions-repeater
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hangtime=0
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authlevel = 0
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[001]
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rxchannel = usrp/127.0.0.1:34002:32002
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duplex = 2
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scheduler=scheduler
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functions = functions-repeater
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hangtime=0
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authlevel = 0
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[002]
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rxchannel = usrp/127.0.0.1:34003:32003
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duplex = 2
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scheduler=scheduler
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functions = functions-repeater
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hangtime=0
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authlevel = 0
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[003]
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rxchannel = usrp/127.0.0.1:34004:32004
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duplex = 2
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scheduler=scheduler
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functions = functions-repeater
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hangtime=0
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authlevel = 0
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[004]
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rxchannel = usrp/127.0.0.1:34005:32005
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duplex = 2
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scheduler=scheduler
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functions = functions-repeater
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hangtime=0
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authlevel = 0
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</pre>
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Continuing the example of five voice channels from above, we define five repeater nodes (channels).  Voice and PTT traffic that 
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is output by asterisk/app_rpt for RF transmission is forwarded to usrp_tx.py (see below) using UDP ports in the 3400x range.
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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.
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The driver invocation in @rpt.conf@ is
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<pre>
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    usrp/HISIP:HISPORT[:MYPORT]       
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    HISIP is the IP address (or FQDN) of the GR app
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    HISPORT is the UDP socket of the GR app
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    MYPORT (optional) is the UDP socket that Asterisk listens on
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             for this channel   
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</pre>
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TIP: You can use the @usrp show@ command to display status information from within the Asterisk CLI.
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TIP: Another handy command is @rpt playback@ to start transmission on a channel.
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h2. Channel Bank Configuration
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Typically the audio links are terminated on channel banks which provide a standard interface to user
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equipment.  Commonly, this equipment places an "offhook" indication on the signalling circuit when it
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wishes to initiate a radio transmission, and signals the end of the transmission by placing the circuit
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in the "onhook" state.  Standard audio transmission levels are defined at the channel bank interface.
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A standard FXS port on the channel bank is defined in @/etc/asterisk/zapata.conf@ with
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<pre>
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signalling=fxo_ls
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immediate=yes
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context = chan1
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channel => 1
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</pre>
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An offhook (PTT) signal from user equipment on the FXS channel bank port 
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(due to the @immmediate=yes@) starts processing in @/etc/asterisk/extensions.conf@:
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<pre>
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[chan1]
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exten => s,1,Dial(local/1@radio/n)
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</pre>
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This jumps to extension "1" in context @radio@ (also in @/etc/asterisk/extensions.conf@)
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<pre>
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[radio]
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exten => 1,1,rpt(000|D)
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exten => 2,1,rpt(001|D)
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exten => 3,1,rpt(002|D)
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exten => 4,1,rpt(003|D)
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exten => 5,1,rpt(004|D)
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</pre>
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So the call resulting from the offhook (PTT) signal is routed to 
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extension "1" in context @[radio]@ where it's connected to the desired
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repeater node (channel).  If the GR app is running it will initiate
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radio transmission.  An onhook signal on the FXS channel bank port
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causes the end of the transmission by ending the asterisk call in
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progress*. The @hangtime=0@ setting in @rpt.conf@ was used to reduce the
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tail delay in this setup.
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*The end of the transmission may be modified however, for example when
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app_rpt appends an "ID" or if a "timeout" occurs.
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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)
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h2. USRP Transmission
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The multi-channel USRP transmitter app currently has some limitations:
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 * Transmit channel spacing is at arbitrary 25 KHz intervals
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 * Analog NBFM mode is not yet supported
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Before running the app you must first determine (example values shown in square brackets - YMMV)
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 * the USRP TX daughterboard ID (A or B) [A]
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 * carrier frequency of the first ("center") TX channel [435.125 MHz]
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 * number of channels to be transmitted [five]
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 * the first UDP port number over which usrp_tx.py receives data from chan_usrp.c [port 34001]
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Example:
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@usrp_tx.py -TA -e -f 435.125e6 -n 5 -p 34001@
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Here we also request a FFT display. With port 34001 as the starting UDP port number and
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five channels, @usrp_tx.py@ listens on ports 34001-34005 for TX traffic coming from chan_usrp.c, as
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configured in @/etc/asterisk/rpt.conf@ (see above).
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h2. Soundcard Transmission
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The soundcard TX program outputs an analog waveform that is suitable for application to the
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modulator stage of an FM transmitter.  It listens on the UDP port for audio frames sent to it
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by asterisk/chan_usrp.  The analog audio thus received is encoded by the soundcard TX app to
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IMBE voice code words, which are in turn assembled into P25 voice frames (LDU1/LDU2's).  The
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resulting 4800 baud symbol stream is then RRC filtered and shaped according to the P25 spec.
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This signal is output to the sound card (and optionally to an on-screen oscilloscope using
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the @-e@ option).
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This app may be run on the same server as asterisk or on a different machine.
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Currently only a single channel is supported (per invocation of soundcard_tx).  If more than
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one TX channel is to be used simultaneously, run a another copy of the app (either on a different
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host or using a different UDP port).
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Syntax:
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@soundcard_tx.py -g 0.3 -p 32001@
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The gain value (@0.3@) would cause the output envelope (ranging from -3 to +3) to be scaled to
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fit within the standard band from -1 to +1.  Values larger than 0.3 may cause clipping and distortion.
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The final output amplitude to the radio can also be adjusted in discrete steps using the audio mixer
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application (e.g., @alsamixer@)
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