FreeCalypso > hg > themwi-system-sw
annotate README @ 191:6ac96217c442
sip-manual-out: add SDP response parsing
author | Mychaela Falconia <falcon@freecalypso.org> |
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date | Fri, 17 Mar 2023 12:07:17 -0800 |
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29 | 1 This Hg repository contains a work-in-progress named Themyscira Wireless system |
2 software. Themyscira Wireless (ThemWi) is an experimental GSM network operated | |
3 by Mother Mychaela of FreeCalypso at a semi-urban/semi-rural location in | |
4 Southern California, USA; this GSM network is operated with Osmocom CNI software | |
5 components, all running on a single Slackware Linux server. ThemWi system sw | |
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6 is a suite of daemon processes and command line tools that run on the same |
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7 machine as all those Osmocom sw components and provide some additional |
29 | 8 functionality that is not provided "out of the box" by Osmocom, most important |
9 of which is outside connectivity to USA PSTN. | |
10 | |
11 We are currently experimenting with using bulkvs.com as our USA PSTN | |
12 connectivity provider. Like most low-cost PSTN connectivity providers, they | |
13 provide the interface to PSTN in the form of a SIP trunk - while I would | |
14 absolutely love to get a traditional TDM trunk instead, with SS7 signaling, | |
15 such a toy would be far beyond my budget, hence I have to settle for SIP. | |
16 Our current status is: | |
17 | |
18 * We have already obtained a block of USA phone numbers (NANP, chosen numbers | |
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19 from an exchange area local to us) from BulkVS. |
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20 |
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21 * These BulkVS-sourced real NANP numbers have been entered as MSISDNs into |
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22 OsmoHLR records for our test SIMs operating on ThemWi GSM. |
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23 |
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24 * We can successfully dial calls from one ThemWi GSM phone to another, with our |
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25 themwi-mncc switch understanding all dialing formats that are considered |
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26 standard for cellular phone networks in USA (full international, or 11 digits |
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27 starting with '1' but no '+', or 10 digits only), as well as our own non- |
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28 standard shorthand dialing method with only 4 digits. |
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29 |
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30 * Whenever someone dials one of our NANP numbers from the outside world, BulkVS |
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31 servers send UDP SIP INVITE packets to our server. Our inbound call gateway |
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32 process is themwi-sip-in; this daemon process listens on UDP port 5060, |
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33 accepts SIP calls from BulkVS (ultimately coming from global worldwide PSTN) |
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34 and turns them into GSM MT calls in MNCC format, going through themwi-mncc |
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35 and ultimately to OsmoMSC. |
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36 |
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37 * Our outbound call gateway is themwi-sip-out, serving as a counterpart to |
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38 themwi-sip-in. All GSM MO calls are initially handled by themwi-mncc, which |
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39 decides (based on the dialed number) whether to switch the call locally |
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40 (calling another ThemWi GSM phone) or to pass it to themwi-sip-out. The |
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41 latter process turns outbound calls into SIP and sends them to BulkVS, and we |
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42 are successfully able to call any PSTN destination anywhere in the +1 country |
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43 code. (themwi-sip-out has provisions for international routes too, but we |
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44 haven't set up any yet.) |
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45 |
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46 * These inbound and outbound calls per the previous two bullet points also |
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47 include fully working voice path, with our themwi-mgw transcoding the two RTP |
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48 streams (one in each direction) between the original GSM 06.10 codec or |
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49 GSM 06.60 EFR codec on the GSM side and G.711 PCMU or PCMA on the PSTN-via-SIP |
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50 side. This voice call gateway includes working DTMF support: START DTMF and |
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51 STOP DTMF commands from GSM phones pass through OsmoMSC, themwi-mncc and |
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52 themwi-sip-{in,out} to themwi-mgw, and the latter process injects in-band DTMF |
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53 tones into the G.711 RTP stream that is otherwise generated by transcoding |
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54 from GSM voice codecs. |
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55 |
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56 The following functionality remains to be implemented: |
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57 |
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58 * The only GSM codecs currently supported are the original FR (GSM 06.10) and |
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59 EFR (GSM 06.60). At some point it would be nice to add support for AMR and |
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60 maybe even HR1, purely for demonstration of its poor voice quality. |
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61 |
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62 * The Presiding High Priestess of Themyscira Wireless has a desire to implement |
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63 in-band GSM 08.62 TFO inside our G.711 RTP interface to outside PSTN. Right |
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64 now if a call is connected between a ThemWi GSM phone on one end and another |
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65 GSM phone on some other network anywhere else in the world, an undesirable |
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66 tandem transcoding takes place: two lossy GSM speech codecs run in tandem, |
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67 one on the ThemWi GSM leg of the call and another on the distant GSM network's |
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68 leg, with a G.711 64 kbps connection in between. If we implement GSM 08.62 |
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69 TFO inside our G.711 RTP interface which we present to the outside world, we |
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70 shall achieve TFO calls at least between ThemWi and any other community GSM |
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71 networks using our software, and maybe even between ThemWi and some legacy |
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72 commercial GSM networks, if any still remain in parts of the world that are |
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73 accessible to us. |
29 | 74 |
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75 * No work whatsoever has been done on SMS as of yet - all of our ThemWi work so |
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76 far has been in support of voice call functionality only. Adding SMS support |
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77 will require 3 major areas of work: |
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78 |
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79 1) We will need to climb the learning curve and assess the current state of SMS |
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80 support in Osmocom CNI software. |
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81 |
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82 2) We'll have to find a USA PSTN connectivity provider who supports P2P SMS: if |
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83 we are operating a GSM network and providing mobile phone service to human |
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84 end users, any SMS usage in such scenario MUST be classified as P2P rather |
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85 than A2P. However, gaining access to P2P SMS in USA telecom environment is |
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86 currently very difficult (every major provider forcibly imposes A2P |
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87 misclassification with no option of P2P correct classification), in stark |
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88 contrast to very easy and cheap access to voice PSTN with NANP phone numbers. |
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89 |
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90 3) When and if we find that necessary USA PSTN connectivity provider with P2P |
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91 SMS provision, we will need to implement whatever software will be needed to |
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92 interconnect it with an OsmoCNI GSM network, likely a custom SMSC. |
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93 |
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94 Differences from osmo-sip-connector |
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95 ----------------------------------- |
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96 |
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97 In the Osmocom community, the "standard" (or generally accepted) way to connect |
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98 a GSM voice network to the outside world is via osmo-sip-connector, an Osmocom |
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99 process that connects to OsmoMSC's MNCC socket on one end and talks SIP on the |
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100 other end. Our combination of themwi-mncc, themwi-sip-in and themwi-sip-out |
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101 effectively takes the place of osmo-sip-connector. Here are the principal ways |
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102 in which our solution differs from osmo-sip-connector: |
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103 |
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104 * o-s-c is designed to connect to a local instance of a SIP PBX such as Asterisk |
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105 or FreeSWITCH, as opposed to interfacing directly to an outside SIP trunk |
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106 from/to a PSTN-via-SIP connectivity provider. themwi-system-sw is different |
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107 in this regard: we do NOT use Asterisk or FreeSWITCH or any other similar |
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108 software of "spaceship" complexity, instead our themwi-sip-in and |
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109 themwi-sip-out processes interface directly to our PSTN-via-SIP connectivity |
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110 provider. |
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111 |
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112 * o-s-c has no internal call switching function for calls from one local GSM |
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113 phone to another, instead such switching is punted to the required Asterisk |
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114 or FreeSWITCH etc. With o-s-c, the calling phone's MO call is converted to |
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115 SIP, then Asterisk or other PBX hairpins it back to o-s-c, and then o-s-c |
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116 handles the destination call leg as a separate conversion from SIP back to |
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117 GSM MNCC. In our solution such local calls are switched internally inside |
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118 themwi-mncc, staying native within GSM MNCC land and never turning into SIP. |
29 | 119 |
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120 * o-s-c is based on Sofia-SIP, which in turn uses glib - a very unpleasant |
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121 dependency in this Mother's opinion. In contrast, our implementation of SIP |
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122 is 100% from scratch, written in plain C in the traditional Falconian coding |
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123 style. |
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124 |
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125 The need for RTP voice transcoding |
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126 ---------------------------------- |
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127 |
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128 In the context of GSM voice codecs, the term "transcoding" is used in two |
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129 significantly different meanings: |
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130 |
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131 * Transcoding from one lossy GSM codec to another effectively constitutes two |
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132 lossy speech codecs running in tandem, and is a highly undesirable condition. |
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133 |
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134 * Running a single GSM codec (not two in tandem), decoding from GSM to G.711 in |
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135 one direction and encoding from G.711 in the other direction, is a standard |
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136 required function for traditional voice gateways between GSM and PSTN. |
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137 |
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138 In themwi-system-sw, we need to do transcoding in the second sense above. |
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139 BulkVS SIP call interface to PSTN does not support any of GSM codecs, they only |
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140 support G.711 and G.729, and the same situation is expected to hold with other |
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141 PSTN-via-SIP connectivity providers. We certainly don't want to use G.729 - we |
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142 don't want to run two lossy speech codecs in tandem, first GSM and then G.729 - |
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143 hence the only codecs we speak on the PSTN-via-SIP side of our gateway are PCMU |
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144 and PCMA. Therefore, we need to perform RTP transcoding in our themwi-mgw, |
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145 very similar to a traditional GSM TRAU. |
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147 On the GSM side, the two codecs of most interest to us at the present time are |
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148 the original FR and EFR - hence they will be the first to be supported. Note |
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149 the big difference from other Osmocom-using GSM community networks which |
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150 typically prefer or even strictly require AMR instead! Our reasons for focusing |
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151 on FR and EFR instead of AMR are: |
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152 |
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153 * Our OsmoBSC time slot configuration is full rate channels only, no half rate |
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154 channels. HR channels are needed only for greater capacity of simultaneous |
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155 calls, but with the total number of people *on the planet* who actively want |
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156 GSM/2G as opposed to LTE or 5G being no more than maybe 10, the thought of |
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157 exceeding the limit of 6 simultaneous call legs per cell (meaning 6 separate |
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158 GSM phone handsets talking *at the same time*) is preposterous. |
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160 * Without any HR channels in OsmoBSC config, AMR means AMR-FR specifically, not |
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161 AMR-HR. The highest level of AMR-FR is identical with EFR - thus if we |
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162 support EFR, do we really need AMR? |
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163 |
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164 * The whole point of Themyscira Wireless is to provide service to *vintage* |
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165 mobile phones. Our current collection of vintage phones includes models that |
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166 only support FR1 and EFR (Ericsson I888, Nokia 5190 and 6190), as well as |
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167 Calypso C05 which supports FR1, EFR and HR1, but not AMR. |
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168 |
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169 * EFR is desirable because it gives better voice quality than FR1, but we must |
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170 support FR1 too, so we can serve the very oldest of phones which support only |
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171 FR1 and nothing else. |
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172 |
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173 Voice codec restriction (forcing GSM phones to use EFR instead of AMR, or |
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174 forcing all the way down to FR1) is done in OsmoBSC config, with codec-list |
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175 setting under 'msc 0'. |