RepeaterPage » History » Version 8
matt, 10/26/2017 12:55 AM
1 | 2 | matt | h1. SUMMARY |
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2 | 1 | zecke | |
3 | * Simultaneous multi-channel transmission/reception |
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4 | * Uses USRP and/or conventional receiver/transmitter hardware |
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5 | * Generic interface to Asterisk via app_rpt (full VOIP, radio control, and standard repeater functions) |
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6 | * Features: P25, analog NBFM with CTCSS |
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7 | * Expected to be fully "ROIP" [Radio Over IP] compatible |
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8 | |||
9 | 2 | matt | h2. Multi-channel reception using the USRP |
10 | 1 | zecke | |
11 | Assume we want to receive simultaneously the four signals shown below; |
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12 | one conventional (analog) FM voice channel plus three P25 |
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13 | signals. The two P25 voice channels are to be IMBE-decoded whereas the |
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14 | P25 data channel is to be sent to Wireshark after decoding. For all three voice |
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15 | channels, we want to forward the received PTT and audio info to Asterisk app_rpt. |
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16 | The received PTT [Push To Talk, or "keyed"] signal is a bit that ideally tracks |
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17 | the state of the PTT key at transmitter, indicating the presence or absence of the |
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18 | received signal. |
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19 | |||
20 | 8 | matt | h2. Spectrum plot |
21 | |||
22 | 7 | matt | !spectrum.png! |
23 | 1 | zecke | |
24 | Fig. 1 - Spectrum of repeater input band (sample; diagram not to scale) |
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25 | |||
26 | For now, it's necessary to edit the source code file manually to specify |
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27 | the list of channels/modes to be received (file usrp_rx.py): |
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28 | |||
29 | 3 | matt | <pre> |
30 | 1 | zecke | channels = [ |
31 | {'freq':435.200e6, 'mode':'c4fm', 'port':32001}, |
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32 | {'freq':435.350e6, 'mode':'fm', 'port':32002, 'ctcss':97.4}, |
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33 | {'freq':435.600e6, 'mode':'cqpsk', 'port':23456, 'wireshark':1}, |
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34 | {'freq':435.775e6, 'mode':'cqpsk', 'port':32003} |
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35 | ] |
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36 | 3 | matt | </pre> |
37 | 1 | zecke | |
38 | Individual channels are defined one per line; note that all definition lines |
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39 | except the last must end with a comma. |
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40 | |||
41 | We choose a frequency somewhere close to the center of this band. |
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42 | which will set the USRP's nominal receive frequency; this must also |
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43 | be manually set in the source file: |
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44 | |||
45 | 3 | matt | <pre> |
46 | 1 | zecke | center_freq = 435.500e6 |
47 | 3 | matt | </pre> |
48 | 1 | zecke | |
49 | Before running the receiver app, we |
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50 | * measure the current calibration error value (I use kalibrate) |
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51 | * determine the optimum USRP receiver gain value |
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52 | The values used in this example are +1234 and 35, respectively. |
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53 | |||
54 | We're now ready to start the receiver: |
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55 | |||
56 | 3 | matt | <pre> |
57 | 1 | zecke | usrp_rx.py -RA -c 1234 -H 127.0.0.1 -g 35 -d 25 |
58 | 3 | matt | </pre> |
59 | 1 | zecke | |
60 | The receiver continuously monitors all four channels. For each of the three |
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61 | voice channels, the audio and the PTT info ("key up" and "key down" events) |
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62 | are forwarded to asterisk app_rpt over separate UDP channels. |
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63 | |||
64 | = P25 and/or analog NBFM Reception using a discriminator-tapped receiver = |
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65 | |||
66 | One or more disc-tapped conventional receivers may be used at the same time, and can |
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67 | coexist with one or more USRP's. |
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68 | |||
69 | An audio cable is connected between the disc-tap point in the receiver and the PC soundcard. |
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70 | |||
71 | Single-channel reception is possible using disctap_rx.py. |
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72 | The app dynmically auto-detects the modulation type (P25 or analog NBFM). |
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73 | |||
74 | Audio and PTT events are forwarded to asterisk app_rpt over two separate UDP |
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75 | channels, depending on modulation type: |
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76 | * received P25 audio is sent to asterisk on UDP port 32004 |
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77 | * when analog NBFM is received with the proper CTCSS tone (97.4 Hz), port 32005 is used |
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78 | |||
79 | 3 | matt | <pre> |
80 | 1 | zecke | disctap_rx.py -i -A 0.05 -c 97.4 -H 127.0.0.1 -p 32004 -g 35 -d 25 |
81 | 3 | matt | </pre> |
82 | 1 | zecke | |
83 | The -g (gain) parameter is used to set the proper audio gain level. See |
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84 | the hardware page for further guidance - this value is important for achieving |
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85 | correct operation. |
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86 | |||
87 | 3 | matt | h2. Asterisk and app_rpt |
88 | 1 | zecke | |
89 | For all voice modes (IMBE and analog FM) the audio is transmitted |
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90 | as frames over the UDP channel to and from Asterisk in the standard native audio format: |
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91 | * 50 frames per second |
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92 | * 160 audio samples per frame |
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93 | * 8000 samples per second |
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94 | * signed |
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95 | * 16-bit |
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96 | * linear |
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97 | |||
98 | 3 | matt | h3. Build asterisk with app_rpt |
99 | 1 | zecke | |
100 | As a prereq make sure the Linux kernel headers are installed, as zaptel installs kernel modules. |
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101 | |||
102 | First obtain and unpack the sources |
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103 | |||
104 | 3 | matt | - http://ohnosec.org/drupal/node/6 [use @svn checkout@ to grab the "Asterisk Sources"] |
105 | 1 | zecke | |
106 | 3 | matt | Locate the toplevel directory - this should have several subdirectories including @zaptel@ and @asterisk@ . |
107 | 1 | zecke | |
108 | 3 | matt | Then locate the subdirectory named @asterisk/channels@ , |
109 | 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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110 | 1 | zecke | |
111 | Now, build and install asterisk and app_rpt |
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112 | |||
113 | 3 | matt | @cd@ to the toplevel directory |
114 | |||
115 | @cd zaptel && ./configure && make && sudo make install@ |
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116 | |||
117 | 1 | zecke | and then (again from the toplevel directory) |
118 | |||
119 | 3 | matt | @cd asterisk && ./configure && make && sudo make install@ |
120 | |||
121 | |||
122 | 1 | zecke | For a first time installation of asterisk there are also other steps such as config file installation. |
123 | 3 | matt | Before setting up @rpt.conf@ you must first set up asterisk itself - see for example |
124 | 1 | zecke | |
125 | http://www.voip-info.org/wiki/view/Asterisk+quickstart |
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126 | |||
127 | 3 | matt | h2. Configuration |
128 | 1 | zecke | |
129 | 3 | matt | We define five repeater nodes in @/etc/asterisk/rpt.conf@ |
130 | 1 | zecke | |
131 | 3 | matt | <pre> |
132 | 1 | zecke | [000] |
133 | rxchannel = usrp/127.0.0.1:34001:32001 |
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134 | duplex = 2 |
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135 | scheduler=scheduler |
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136 | functions = functions-repeater |
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137 | hangtime=0 |
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138 | authlevel = 0 |
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139 | |||
140 | [001] |
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141 | rxchannel = usrp/127.0.0.1:34002:32002 |
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142 | duplex = 2 |
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143 | scheduler=scheduler |
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144 | functions = functions-repeater |
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145 | hangtime=0 |
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146 | authlevel = 0 |
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147 | |||
148 | [002] |
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149 | rxchannel = usrp/127.0.0.1:34003:32003 |
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150 | duplex = 2 |
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151 | scheduler=scheduler |
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152 | functions = functions-repeater |
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153 | hangtime=0 |
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154 | authlevel = 0 |
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155 | |||
156 | [003] |
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157 | rxchannel = usrp/127.0.0.1:34004:32004 |
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158 | duplex = 2 |
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159 | scheduler=scheduler |
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160 | functions = functions-repeater |
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161 | hangtime=0 |
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162 | authlevel = 0 |
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163 | |||
164 | [004] |
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165 | rxchannel = usrp/127.0.0.1:34005:32005 |
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166 | duplex = 2 |
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167 | scheduler=scheduler |
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168 | functions = functions-repeater |
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169 | hangtime=0 |
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170 | authlevel = 0 |
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171 | 3 | matt | </pre> |
172 | 1 | zecke | |
173 | Continuing the example of five voice channels from above, we define five repeater nodes (channels). Voice and PTT traffic that |
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174 | 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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175 | 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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176 | |||
177 | 3 | matt | The driver invocation in @rpt.conf@ is |
178 | <pre> |
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179 | 1 | zecke | usrp/HISIP:HISPORT[:MYPORT] |
180 | HISIP is the IP address (or FQDN) of the GR app |
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181 | HISPORT is the UDP socket of the GR app |
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182 | MYPORT (optional) is the UDP socket that Asterisk listens on |
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183 | for this channel |
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184 | 3 | matt | </pre> |
185 | 1 | zecke | |
186 | 4 | matt | TIP: You can use the @usrp show@ command to display status information from within the Asterisk CLI. |
187 | 1 | zecke | |
188 | 3 | matt | TIP: Another handy command is @rpt playback@ to start transmission on a channel. |
189 | 1 | zecke | |
190 | 3 | matt | h2. Channel Bank Configuration |
191 | 1 | zecke | |
192 | Typically the audio links are terminated on channel banks which provide a standard interface to user |
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193 | equipment. Commonly, this equipment places an "offhook" indication on the signalling circuit when it |
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194 | wishes to initiate a radio transmission, and signals the end of the transmission by placing the circuit |
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195 | in the "onhook" state. Standard audio transmission levels are defined at the channel bank interface. |
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196 | |||
197 | 3 | matt | A standard FXS port on the channel bank is defined in @/etc/asterisk/zapata.conf@ with |
198 | <pre> |
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199 | 1 | zecke | signalling=fxo_ls |
200 | immediate=yes |
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201 | context = chan1 |
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202 | channel => 1 |
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203 | 3 | matt | </pre> |
204 | 1 | zecke | |
205 | An offhook (PTT) signal from user equipment on the FXS channel bank port |
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206 | 3 | matt | (due to the @immmediate=yes@) starts processing in @/etc/asterisk/extensions.conf@: |
207 | 1 | zecke | |
208 | 3 | matt | <pre> |
209 | 1 | zecke | [chan1] |
210 | exten => s,1,Dial(local/1@radio/n) |
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211 | 3 | matt | </pre> |
212 | This jumps to extension "1" in context @radio@ (also in @/etc/asterisk/extensions.conf@) |
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213 | <pre> |
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214 | 1 | zecke | [radio] |
215 | exten => 1,1,rpt(000|D) |
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216 | exten => 2,1,rpt(001|D) |
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217 | exten => 3,1,rpt(002|D) |
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218 | exten => 4,1,rpt(003|D) |
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219 | exten => 5,1,rpt(004|D) |
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220 | 3 | matt | </pre> |
221 | 1 | zecke | |
222 | So the call resulting from the offhook (PTT) signal is routed to |
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223 | 3 | matt | extension "1" in context @[radio]@ where it's connected to the desired |
224 | 1 | zecke | repeater node (channel). If the GR app is running it will initiate |
225 | radio transmission. An onhook signal on the FXS channel bank port |
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226 | causes the end of the transmission by ending the asterisk call in |
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227 | 3 | matt | progress*. The @hangtime=0@ setting in @rpt.conf@ was used to reduce the |
228 | 1 | zecke | tail delay in this setup. |
229 | |||
230 | *The end of the transmission may be modified however, for example when |
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231 | app_rpt appends an "ID" or if a "timeout" occurs. |
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232 | |||
233 | 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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234 | |||
235 | 3 | matt | h2. USRP Transmission |
236 | 1 | zecke | |
237 | The multi-channel USRP transmitter app currently has some limitations: |
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238 | * Transmit channel spacing is at arbitrary 25 KHz intervals |
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239 | * Analog NBFM mode is not yet supported |
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240 | Before running the app you must first determine (example values shown in square brackets - YMMV) |
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241 | * the USRP TX daughterboard ID (A or B) [A] |
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242 | * carrier frequency of the first ("center") TX channel [435.125 MHz] |
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243 | * number of channels to be transmitted [five] |
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244 | * the first UDP port number over which usrp_tx.py receives data from chan_usrp.c [port 34001] |
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245 | Example: |
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246 | 3 | matt | |
247 | @usrp_tx.py -TA -e -f 435.125e6 -n 5 -p 34001@ |
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248 | |||
249 | 1 | zecke | Here we also request a FFT display. With port 34001 as the starting UDP port number and |
250 | 3 | matt | five channels, @usrp_tx.py@ listens on ports 34001-34005 for TX traffic coming from chan_usrp.c, as |
251 | configured in @/etc/asterisk/rpt.conf@ (see above). |
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252 | 1 | zecke | |
253 | 3 | matt | h2. Soundcard Transmission |
254 | 1 | zecke | |
255 | The soundcard TX program outputs an analog waveform that is suitable for application to the |
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256 | modulator stage of an FM transmitter. It listens on the UDP port for audio frames sent to it |
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257 | by asterisk/chan_usrp. The analog audio thus received is encoded by the soundcard TX app to |
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258 | IMBE voice code words, which are in turn assembled into P25 voice frames (LDU1/LDU2's). The |
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259 | resulting 4800 baud symbol stream is then RRC filtered and shaped according to the P25 spec. |
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260 | This signal is output to the sound card (and optionally to an on-screen oscilloscope using |
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261 | 3 | matt | the @-e@ option). |
262 | 1 | zecke | |
263 | This app may be run on the same server as asterisk or on a different machine. |
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264 | |||
265 | Currently only a single channel is supported (per invocation of soundcard_tx). If more than |
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266 | one TX channel is to be used simultaneously, run a another copy of the app (either on a different |
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267 | host or using a different UDP port). |
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268 | |||
269 | Syntax: |
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270 | |||
271 | 5 | matt | @soundcard_tx.py -g 0.3 -p 32001@ |
272 | 3 | matt | |
273 | |||
274 | The gain value (@0.3@) would cause the output envelope (ranging from -3 to +3) to be scaled to |
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275 | 1 | zecke | fit within the standard band from -1 to +1. Values larger than 0.3 may cause clipping and distortion. |
276 | |||
277 | The final output amplitude to the radio can also be adjusted in discrete steps using the audio mixer |
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278 | 3 | matt | application (e.g., @alsamixer@) |