FreeCalypso > hg > themwi-system-sw

annotate README @ 196:eac3e0b6ce02

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sip-manual-out: (hopefully) fix RTP output timing
author Mychaela Falconia <falcon@freecalypso.org>
date Fri, 17 Mar 2023 17:59:41 -0800
parents b259e2722485
children
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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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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 * 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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56 The following functionality remains to be implemented:
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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.
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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.
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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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146
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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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159
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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'.