RepeaterPage » History » Version 12
max, 04/22/2017 04:04 PM
1 | 1 | max | |
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2 | 4 | max | = SUMMARY = |
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 | 1 | max | = Multi-channel reception using the USRP = |
10 | |||
11 | 6 | max | Assume we want to receive simultaneously the four signals shown below; |
12 | 1 | max | one conventional (analog) FM voice channel plus three P25 |
13 | signals. The two P25 voice channels are to be IMBE-decoded whereas the |
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14 | 4 | max | P25 data channel is to be sent to Wireshark after decoding. For all three voice |
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 | 1 | max | |
20 | 3 | max | [[Image(sa.png)]] |
21 | |||
22 | 1 | max | Fig. 1 - Spectrum of repeater input band (sample; diagram not to scale) |
23 | |||
24 | For now, it's necessary to edit the source code file manually to specify |
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25 | the list of channels/modes to be received (file usrp_rx.py): |
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26 | |||
27 | {{{ |
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28 | channels = [ |
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29 | {'freq':435.200e6, 'mode':'c4fm', 'port':32001}, |
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30 | {'freq':435.350e6, 'mode':'fm', 'port':32002, 'ctcss':97.4}, |
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31 | {'freq':435.600e6, 'mode':'cqpsk', 'port':23456, 'wireshark':1}, |
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32 | {'freq':435.775e6, 'mode':'cqpsk', 'port':32003} |
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33 | ] |
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34 | }}} |
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35 | |||
36 | Individual channels are defined one per line; note that all definition lines |
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37 | except the last must end with a comma. |
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38 | |||
39 | We choose a frequency somewhere close to the center of this band. |
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40 | which will set the USRP's nominal receive frequency; this must also |
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41 | be manually set in the source file: |
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42 | |||
43 | {{{ |
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44 | center_freq = 435.500e6 |
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45 | }}} |
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46 | |||
47 | Before running the receiver app, we |
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48 | * measure the current calibration error value (I use kalibrate) |
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49 | * determine the optimum USRP receiver gain value |
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50 | The values used in this example are +1234 and 35, respectively. |
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51 | |||
52 | We're now ready to start the receiver: |
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53 | |||
54 | {{{ |
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55 | usrp_rx.py -RA -c 1234 -H 127.0.0.1 -g 35 -d 25 |
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56 | }}} |
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57 | |||
58 | The receiver continuously monitors all four channels. For each of the three |
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59 | voice channels, the audio and the PTT info ("key up" and "key down" events) |
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60 | are forwarded to asterisk app_rpt over separate UDP channels. |
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61 | |||
62 | = P25 and/or analog NBFM Reception using a discriminator-tapped receiver = |
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63 | |||
64 | One or more disc-tapped conventional receivers may be used at the same time, and can |
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65 | coexist with one or more USRP's. |
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66 | |||
67 | 4 | max | An audio cable is connected between the disc-tap point in the receiver and the PC soundcard. |
68 | |||
69 | 1 | max | Single-channel reception is possible using disctap_rx.py. |
70 | 4 | max | The app dynmically auto-detects the modulation type (P25 or analog NBFM). |
71 | 1 | max | |
72 | Audio and PTT events are forwarded to asterisk app_rpt over two separate UDP |
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73 | 4 | max | channels, depending on modulation type: |
74 | * received P25 audio is sent to asterisk on UDP port 32004 |
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75 | * when analog NBFM is received with the proper CTCSS tone (97.4 Hz), port 32005 is used |
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76 | 1 | max | |
77 | {{{ |
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78 | 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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79 | }}} |
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80 | |||
81 | The -g (gain) parameter is used to set the proper audio gain level. See |
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82 | the hardware page for further guidance - this value is important for achieving |
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83 | correct operation. |
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84 | 4 | max | |
85 | = Asterisk and app_rpt = |
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86 | |||
87 | For all voice modes (IMBE and analog FM) the audio is transmitted |
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88 | as frames over the UDP channel to and from Asterisk in the standard native audio format: |
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89 | * 50 frames per second |
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90 | * 160 audio samples per frame |
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91 | 6 | max | * 8000 samples per second |
92 | 4 | max | * signed |
93 | * 16-bit |
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94 | * linear |
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95 | |||
96 | 10 | max | == Build asterisk with app_rpt == |
97 | |||
98 | 11 | max | As a prereq make sure the Linux kernel headers are installed, as zaptel installs kernel modules. |
99 | |||
100 | 10 | max | First obtain and unpack the sources |
101 | |||
102 | - http://ohnosec.org/drupal/node/6 [use {{{svn checkout}}} to grab the "Asterisk Sources"] |
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103 | |||
104 | Locate the toplevel directory - this should have several subdirectories including {{{zaptel}}} and {{{asterisk}}} . |
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105 | |||
106 | Then locate the subdirectory named {{{asterisk/channels}}} , |
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107 | 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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108 | |||
109 | Now, build and install asterisk and app_rpt |
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110 | |||
111 | {{{cd}}} to the toplevel directory |
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112 | {{{ |
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113 | cd zaptel && ./configure && make && sudo make install |
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114 | }}} |
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115 | and then (again from the toplevel directory) |
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116 | {{{ |
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117 | cd asterisk && ./configure && make && sudo make install |
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118 | }}} |
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119 | |||
120 | For a first time installation of asterisk there are also other steps such as config file installation. |
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121 | Before setting up {{{rpt.conf}}} you must first set up asterisk itself - see for example |
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122 | |||
123 | http://www.voip-info.org/wiki/view/Asterisk+quickstart |
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124 | |||
125 | 5 | max | == Configuration == |
126 | |||
127 | We define five repeater nodes in {{{/etc/asterisk/rpt.conf}}} |
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128 | |||
129 | {{{ |
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130 | [000] |
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131 | rxchannel = usrp/127.0.0.1:34001:32001 |
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132 | duplex = 2 |
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133 | scheduler=scheduler |
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134 | functions = functions-repeater |
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135 | hangtime=0 |
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136 | authlevel = 0 |
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137 | |||
138 | [001] |
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139 | rxchannel = usrp/127.0.0.1:34002:32002 |
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140 | duplex = 2 |
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141 | scheduler=scheduler |
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142 | functions = functions-repeater |
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143 | hangtime=0 |
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144 | authlevel = 0 |
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145 | |||
146 | [002] |
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147 | rxchannel = usrp/127.0.0.1:34003:32003 |
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148 | duplex = 2 |
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149 | scheduler=scheduler |
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150 | functions = functions-repeater |
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151 | hangtime=0 |
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152 | authlevel = 0 |
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153 | |||
154 | [003] |
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155 | rxchannel = usrp/127.0.0.1:34004:32004 |
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156 | duplex = 2 |
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157 | scheduler=scheduler |
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158 | functions = functions-repeater |
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159 | hangtime=0 |
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160 | authlevel = 0 |
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161 | |||
162 | [004] |
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163 | rxchannel = usrp/127.0.0.1:34005:32005 |
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164 | duplex = 2 |
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165 | scheduler=scheduler |
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166 | functions = functions-repeater |
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167 | hangtime=0 |
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168 | authlevel = 0 |
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169 | }}} |
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170 | |||
171 | Continuing the example of five voice channels from above, we define five repeater nodes (channels). Voice and PTT traffic that |
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172 | 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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173 | 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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174 | |||
175 | The driver invocation in {{{rpt.conf}}} is |
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176 | {{{ |
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177 | usrp/HISIP:HISPORT[:MYPORT] |
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178 | HISIP is the IP address (or FQDN) of the GR app |
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179 | HISPORT is the UDP socket of the GR app |
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180 | MYPORT (optional) is the UDP socket that Asterisk listens on |
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181 | for this channel |
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182 | }}} |
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183 | |||
184 | TIP: You can use the {{{usrp show}}} command to display status information from within the Asterisk CLI. |
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185 | |||
186 | TIP: Another handy command is {{{rpt playback}}} to start transmission on a channel. |
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187 | |||
188 | == Channel Bank Configuration == |
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189 | |||
190 | Typically the audio links are terminated on channel banks which provide a standard interface to user |
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191 | equipment. Commonly, this equipment places an "offhook" indication on the signalling circuit when it |
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192 | wishes to initiate a radio transmission, and signals the end of the transmission by placing the circuit |
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193 | in the "onhook" state. Standard audio transmission levels are defined at the channel bank interface. |
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194 | |||
195 | A standard FXS port on the channel bank is defined in {{{/etc/asterisk/zapata.conf}}} with |
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196 | {{{ |
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197 | signalling=fxo_ls |
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198 | immediate=yes |
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199 | context = chan1 |
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200 | channel => 1 |
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201 | }}} |
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202 | |||
203 | An offhook (PTT) signal from user equipment on the FXS channel bank port |
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204 | (due to the {{{immmediate=yes}}}) starts processing in {{{/etc/asterisk/extensions.conf}}}: |
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205 | |||
206 | {{{ |
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207 | 1 | max | [chan1] |
208 | exten => s,1,Dial(local/1@radio/n) |
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209 | 6 | max | }}} |
210 | This jumps to extension "1" in context {{{radio}}} (also in {{{/etc/asterisk/extensions.conf}}}) |
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211 | {{{ |
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212 | 5 | max | [radio] |
213 | exten => 1,1,rpt(000|D) |
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214 | exten => 2,1,rpt(001|D) |
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215 | exten => 3,1,rpt(002|D) |
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216 | exten => 4,1,rpt(003|D) |
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217 | exten => 5,1,rpt(004|D) |
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218 | }}} |
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219 | |||
220 | So the call resulting from the offhook (PTT) signal is routed to |
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221 | extension "1" in context {{{[radio]}}} where it's connected to the desired |
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222 | repeater node (channel). If the GR app is running it will initiate |
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223 | radio transmission. An onhook signal on the FXS channel bank port |
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224 | causes the end of the transmission by ending the asterisk call in |
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225 | progress*. The {{{hangtime=0}}} setting in {{{rpt.conf}}} was used to reduce the |
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226 | tail delay in this setup. |
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227 | |||
228 | *The end of the transmission may be modified however, for example when |
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229 | 1 | max | app_rpt appends an "ID" or if a "timeout" occurs. |
230 | |||
231 | 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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232 | |||
233 | 6 | max | = USRP Transmission = |
234 | 1 | max | |
235 | 6 | max | The multi-channel USRP transmitter app currently has some limitations: |
236 | * Transmit channel spacing is at arbitrary 25 KHz intervals |
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237 | * Analog NBFM mode is not yet supported |
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238 | Before running the app you must first determine (example values shown in square brackets - YMMV) |
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239 | * the USRP TX daughterboard ID (A or B) [A] |
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240 | * carrier frequency of the first ("center") TX channel [435.125 MHz] |
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241 | * number of channels to be transmitted [five] |
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242 | 7 | max | * the first UDP port number over which usrp_tx.py receives data from chan_usrp.c [port 34001] |
243 | 6 | max | Example: |
244 | {{{ |
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245 | usrp_tx.py -TA -e -f 435.125e6 -n 5 -p 34001 |
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246 | }}} |
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247 | Here we also request a FFT display. With port 34001 as the starting UDP port number and |
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248 | five channels, {{{usrp_tx.py}}} listens on ports 34001-34005 for TX traffic coming from chan_usrp.c, as |
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249 | configured in {{{/etc/asterisk/rpt.conf}}} (see above). |
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250 | 8 | max | |
251 | = Soundcard Transmission = |
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252 | |||
253 | The soundcard TX program outputs an analog waveform that is suitable for application to the |
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254 | modulator stage of an FM transmitter. It listens on the UDP port for audio frames sent to it |
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255 | by asterisk/chan_usrp. The analog audio thus received is encoded by the soundcard TX app to |
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256 | IMBE voice code words, which are in turn assembled into P25 voice frames (LDU1/LDU2's). The |
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257 | resulting 4800 baud symbol stream is then RRC filtered and shaped according to the P25 spec. |
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258 | This signal is output to the sound card (and optionally to an on-screen oscilloscope using |
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259 | the {{{-e}}} option). |
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260 | |||
261 | This app may be run on the same server as asterisk or on a different machine. |
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262 | |||
263 | Currently only a single channel is supported (per invocation of soundcard_tx). If more than |
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264 | one TX channel is to be used simultaneously, run a another copy of the app (either on a different |
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265 | host or using a different UDP port). |
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266 | |||
267 | Syntax: |
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268 | {{{ |
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269 | soundcard_tx.py -g 0.3 -p 32001 |
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270 | }}} |
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271 | |||
272 | 9 | max | The gain value ({{{0.3}}}) would cause the output envelope (ranging from -3 to +3) to be scaled to |
273 | 8 | max | fit within the standard band from -1 to +1. Values larger than 0.3 may cause clipping and distortion. |
274 | |||
275 | The final output amplitude to the radio can also be adjusted in discrete steps using the audio mixer |
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276 | application (e.g., {{{alsamixer}}}) |