FreeCalypso > hg > themwi-system-sw
annotate README @ 131:5c7109183c8c
shorten new program name to themwi-update-outrt
author | Mychaela Falconia <falcon@freecalypso.org> |
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date | Thu, 06 Oct 2022 21:19:59 -0800 |
parents | b7cd66acb123 |
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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 * These inbound calls per the previous bullet point also include fully working |
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38 voice path, with our themwi-mgw transcoding the two RTP streams (one in each |
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39 direction) between the original GSM 06.10 codec on the GSM side and G.711 |
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40 PCMU or PCMA on the PSTN-via-SIP side. This voice call gateway includes |
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41 working DTMF support: START DTMF and STOP DTMF commands from GSM phones pass |
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42 through OsmoMSC, themwi-mncc and themwi-sip-in to themwi-mgw, and the latter |
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43 process injects in-band DTMF tones into the G.711 RTP stream that is otherwise |
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44 generated by transcoding from GSM voice codecs. |
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45 |
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46 The following functionality remains to be implemented: |
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47 |
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48 * As a counterpart to themwi-sip-in, there will be another process named |
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49 themwi-sip-out that will serve as a gateway for outbound calls, going from |
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50 GSM MO MNCC to outside PSTN via SIP. The outbound SIP call functional part |
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51 is already implemented in test prototype form in sip-manual-out. |
29 | 52 |
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53 * Right now themwi-mgw supports only the original FR1 codec (GSM 06.10) on the |
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54 GSM side; the Mother's desire is to also support EFR codec as a high priority, |
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55 and maybe some time later AMR as a lower priority. |
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56 |
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57 Differences from osmo-sip-connector |
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58 ----------------------------------- |
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59 |
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60 In the Osmocom community, the "standard" (or generally accepted) way to connect |
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61 a GSM voice network to the outside world is via osmo-sip-connector, an Osmocom |
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62 process that connects to OsmoMSC's MNCC socket on one end and talks SIP on the |
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63 other end. Our combination of themwi-mncc, themwi-sip-in and themwi-sip-out |
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64 effectively takes the place of osmo-sip-connector. Here are the principal ways |
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65 in which our solution differs from osmo-sip-connector: |
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66 |
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67 * o-s-c is designed to connect to a local instance of a SIP PBX such as Asterisk |
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68 or FreeSWITCH, as opposed to interfacing directly to an outside SIP trunk |
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69 from/to a PSTN-via-SIP connectivity provider. themwi-system-sw is different |
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70 in this regard: we do NOT use Asterisk or FreeSWITCH or any other similar |
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71 software of "spaceship" complexity, instead our themwi-sip-in and |
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72 themwi-sip-out processes interface directly to our PSTN-via-SIP connectivity |
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73 provider. |
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74 |
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75 * o-s-c has no internal call switching function for calls from one local GSM |
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76 phone to another, instead such switching is punted to the required Asterisk |
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77 or FreeSWITCH etc. With o-s-c, the calling phone's MO call is converted to |
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78 SIP, then Asterisk or other PBX hairpins it back to o-s-c, and then o-s-c |
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79 handles the destination call leg as a separate conversion from SIP back to |
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80 GSM MNCC. In our solution such local calls are switched internally inside |
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81 themwi-mncc, staying native within GSM MNCC land and never turning into SIP. |
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83 * o-s-c is based on Sofia-SIP, which in turn uses glib - a very unpleasant |
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84 dependency in this Mother's opinion. In contrast, our implementation of SIP |
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85 is 100% from scratch, written in plain C in the traditional Falconian coding |
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86 style. |
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87 |
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88 The need for RTP voice transcoding |
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89 ---------------------------------- |
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90 |
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91 In the context of GSM voice codecs, the term "transcoding" is used in two |
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92 significantly different meanings: |
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93 |
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94 * Transcoding from one lossy GSM codec to another effectively constitutes two |
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95 lossy speech codecs running in tandem, and is a highly undesirable condition. |
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96 |
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97 * Running a single GSM codec (not two in tandem), decoding from GSM to G.711 in |
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98 one direction and encoding from G.711 in the other direction, is a standard |
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99 required function for traditional voice gateways between GSM and PSTN. |
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100 |
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101 In themwi-system-sw, we need to do transcoding in the second sense above. |
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102 BulkVS SIP call interface to PSTN does not support any of GSM codecs, they only |
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103 support G.711 and G.729, and the same situation is expected to hold with other |
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104 PSTN-via-SIP connectivity providers. We certainly don't want to use G.729 - we |
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105 don't want to run two lossy speech codecs in tandem, first GSM and then G.729 - |
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106 hence the only codecs we speak on the PSTN-via-SIP side of our gateway are PCMU |
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107 and PCMA. Therefore, we need to perform RTP transcoding in our themwi-mgw, |
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108 very similar to a traditional GSM TRAU. |
29 | 109 |
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110 On the GSM side, the two codecs of most interest to us at the present time are |
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111 the original FR and EFR - hence they will be the first to be supported. Note |
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112 the big difference from other Osmocom-using GSM community networks which |
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113 typically prefer or even strictly require AMR instead! Our reasons for focusing |
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114 on FR and EFR instead of AMR are: |
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115 |
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116 * Our OsmoBSC time slot configuration is full rate channels only, no half rate |
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117 channels. HR channels are needed only for greater capacity of simultaneous |
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118 calls, but with the total number of people *on the planet* who actively want |
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119 GSM/2G as opposed to LTE or 5G being no more than maybe 10, the thought of |
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120 exceeding the limit of 6 simultaneous call legs per cell (meaning 6 separate |
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121 GSM phone handsets talking *at the same time*) is preposterous. |
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123 * Without any HR channels in OsmoBSC config, AMR means AMR-FR specifically, not |
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124 AMR-HR. The highest level of AMR-FR is identical with EFR - thus if we |
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125 support EFR, do we really need AMR? |
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126 |
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127 * The whole point of Themyscira Wireless is to provide service to *vintage* |
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128 mobile phones. Our current collection of vintage phones includes models that |
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129 only support FR1 and EFR (Ericsson I888, Nokia 5190 and 6190), as well as |
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130 Calypso C05 which supports FR1, EFR and HR1, but not AMR. |
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131 |
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132 * EFR is desirable because it gives better voice quality than FR1, but we must |
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133 support FR1 too, so we can serve the very oldest of phones which support only |
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134 FR1 and nothing else. |
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135 |
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136 Voice codec restriction (forcing GSM phones to use EFR instead of AMR, or |
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137 forcing all the way down to FR1) is done in OsmoBSC config, with codec-list |
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138 setting under 'msc 0'. |