FreeCalypso > hg > themwi-system-sw
annotate README @ 94:2c22b40408fb
sip-in BYE UAC: use new libsip function for response ident
author | Mychaela Falconia <falcon@freecalypso.org> |
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date | Sat, 24 Sep 2022 15:14:50 -0800 |
parents | 97317ede320a |
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rev | line source |
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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 The following functionality remains to be implemented: |
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38 |
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39 * As a counterpart to themwi-sip-in, there will be another process named |
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40 themwi-sip-out that will serve as a gateway for outbound calls, going from |
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41 GSM MO MNCC to outside PSTN via SIP. The outbound SIP call functional part |
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42 is already implemented in test prototype form in sip-manual-out. |
29 | 43 |
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44 * themwi-mgw will be our transcoding RTP bridge, speaking GSM codecs (FR and |
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45 EFR are currently of most interest) on the side toward Osmocom components and |
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46 G.711 (PCMU or PCMA) on the PSTN side. Right now themwi-mgw is a working |
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47 skeleton that allocates endpoints with RTP & RTCP UDP port pairs, but doesn't |
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48 pass any traffic yet. |
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49 |
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50 Differences from osmo-sip-connector |
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51 ----------------------------------- |
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52 |
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53 In the Osmocom community, the "standard" (or generally accepted) way to connect |
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54 a GSM voice network to the outside world is via osmo-sip-connector, an Osmocom |
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55 process that connects to OsmoMSC's MNCC socket on one end and talks SIP on the |
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56 other end. Our combination of themwi-mncc, themwi-sip-in and themwi-sip-out |
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57 effectively takes the place of osmo-sip-connector. Here are the principal ways |
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58 in which our solution differs from osmo-sip-connector: |
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59 |
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60 * o-s-c is designed to connect to a local instance of a SIP PBX such as Asterisk |
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61 or FreeSWITCH, as opposed to interfacing directly to an outside SIP trunk |
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62 from/to a PSTN-via-SIP connectivity provider. themwi-system-sw is different |
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63 in this regard: we do NOT use Asterisk or FreeSWITCH or any other similar |
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64 software of "spaceship" complexity, instead our themwi-sip-in and |
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65 themwi-sip-out processes interface directly to our PSTN-via-SIP connectivity |
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66 provider. |
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67 |
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68 * o-s-c has no internal call switching function for calls from one local GSM |
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69 phone to another, instead such switching is punted to the required Asterisk |
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70 or FreeSWITCH etc. With o-s-c, the calling phone's MO call is converted to |
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71 SIP, then Asterisk or other PBX hairpins it back to o-s-c, and then o-s-c |
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72 handles the destination call leg as a separate conversion from SIP back to |
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73 GSM MNCC. In our solution such local calls are switched internally inside |
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74 themwi-mncc, staying native within GSM MNCC land and never turning into SIP. |
29 | 75 |
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76 * o-s-c is based on Sofia-SIP, which in turn uses glib - a very unpleasant |
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77 dependency in this Mother's opinion. In contrast, our implementation of SIP |
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78 is 100% from scratch, written in plain C in the traditional Falconian coding |
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79 style. |
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80 |
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81 The need for RTP voice transcoding |
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82 ---------------------------------- |
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83 |
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84 In the context of GSM voice codecs, the term "transcoding" is used in two |
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85 significantly different meanings: |
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86 |
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87 * Transcoding from one lossy GSM codec to another effectively constitutes two |
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88 lossy speech codecs running in tandem, and is a highly undesirable condition. |
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89 |
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90 * Running a single GSM codec (not two in tandem), decoding from GSM to G.711 in |
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91 one direction and encoding from G.711 in the other direction, is a standard |
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92 required function for traditional voice gateways between GSM and PSTN. |
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93 |
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94 In themwi-system-sw, we need to do transcoding in the second sense above. |
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95 BulkVS SIP call interface to PSTN does not support any of GSM codecs, they only |
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96 support G.711 and G.729, and the same situation is expected to hold with other |
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97 PSTN-via-SIP connectivity providers. We certainly don't want to use G.729 - we |
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98 don't want to run two lossy speech codecs in tandem, first GSM and then G.729 - |
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99 hence the only codecs we speak on the PSTN-via-SIP side of our gateway are PCMU |
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100 and PCMA. Therefore, we need to perform RTP transcoding in our themwi-mgw, |
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101 very similar to a traditional GSM TRAU. |
29 | 102 |
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103 On the GSM side, the two codecs of most interest to us at the present time are |
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104 the original FR and EFR - hence they will be the first to be supported. Note |
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105 the big difference from other Osmocom-using GSM community networks which |
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106 typically prefer or even strictly require AMR instead! Our reasons for focusing |
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107 on FR and EFR instead of AMR are: |
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108 |
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109 * Our OsmoBSC time slot configuration is full rate channels only, no half rate |
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110 channels. HR channels are needed only for greater capacity of simultaneous |
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111 calls, but with the total number of people *on the planet* who actively want |
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112 GSM/2G as opposed to LTE or 5G being no more than maybe 10, the thought of |
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113 exceeding the limit of 6 simultaneous call legs per cell (meaning 6 separate |
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114 GSM phone handsets talking *at the same time*) is preposterous. |
29 | 115 |
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116 * Without any HR channels in OsmoBSC config, AMR means AMR-FR specifically, not |
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117 AMR-HR. The highest level of AMR-FR is identical with EFR - thus if we |
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118 support EFR, do we really need AMR? |
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119 |
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120 * The whole point of Themyscira Wireless is to provide service to *vintage* |
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121 mobile phones. Our current collection of vintage phones includes models that |
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122 only support FR1 and EFR (Ericsson I888, Nokia 5190) and Calypso C05 which |
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123 supports FR1, EFR and HR1, but not AMR. |
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124 |
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125 * EFR is desirable because it gives better voice quality than FR1, but we must |
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126 support FR1 too, so we can serve the very oldest of phones which support only |
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127 FR1 and nothing else. |
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128 |
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29
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changeset
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129 Voice codec restriction (forcing GSM phones to use EFR instead of AMR, or |
97317ede320a
README: update for current status
Mychaela Falconia <falcon@freecalypso.org>
parents:
29
diff
changeset
|
130 forcing all the way down to FR1) is done in OsmoBSC config, with codec-list |
97317ede320a
README: update for current status
Mychaela Falconia <falcon@freecalypso.org>
parents:
29
diff
changeset
|
131 setting under 'msc 0'. |