FreeCalypso > hg > themwi-system-sw

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sip-out: handle incoming BYE as UAS
author Mychaela Falconia <falcon@freecalypso.org>
date Wed, 12 Oct 2022 14:45:31 -0800
parents b7cd66acb123
children b259e2722485
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1 This Hg repository contains a work-in-progress named Themyscira Wireless system
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2 software. Themyscira Wireless (ThemWi) is an experimental GSM network operated
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3 by Mother Mychaela of FreeCalypso at a semi-urban/semi-rural location in
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4 Southern California, USA; this GSM network is operated with Osmocom CNI software
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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
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8 functionality that is not provided "out of the box" by Osmocom, most important
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9 of which is outside connectivity to USA PSTN.
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10
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11 We are currently experimenting with using bulkvs.com as our USA PSTN
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12 connectivity provider. Like most low-cost PSTN connectivity providers, they
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13 provide the interface to PSTN in the form of a SIP trunk - while I would
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14 absolutely love to get a traditional TDM trunk instead, with SS7 signaling,
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15 such a toy would be far beyond my budget, hence I have to settle for SIP.
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16 Our current status is:
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17
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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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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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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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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.
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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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82
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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.
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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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Mychaela Falconia <falcon@freecalypso.org>
parents: 29
diff changeset
132 * EFR is desirable because it gives better voice quality than FR1, but we must
97317ede320a README: update for current status
Mychaela Falconia <falcon@freecalypso.org>
parents: 29
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133 support FR1 too, so we can serve the very oldest of phones which support only
97317ede320a README: update for current status
Mychaela Falconia <falcon@freecalypso.org>
parents: 29
diff changeset
134 FR1 and nothing else.
97317ede320a README: update for current status
Mychaela Falconia <falcon@freecalypso.org>
parents: 29
diff changeset
135
97317ede320a README: update for current status
Mychaela Falconia <falcon@freecalypso.org>
parents: 29
diff changeset
136 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
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137 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
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138 setting under 'msc 0'.