FreeCalypso > hg > themwi-system-sw
comparison README @ 178:b259e2722485
README: update for current status
author | Mychaela Falconia <falcon@freecalypso.org> |
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date | Thu, 09 Mar 2023 13:08:31 -0800 |
parents | b7cd66acb123 |
children |
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177:a851bde42d82 | 178:b259e2722485 |
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32 process is themwi-sip-in; this daemon process listens on UDP port 5060, | 32 process is themwi-sip-in; this daemon process listens on UDP port 5060, |
33 accepts SIP calls from BulkVS (ultimately coming from global worldwide PSTN) | 33 accepts SIP calls from BulkVS (ultimately coming from global worldwide PSTN) |
34 and turns them into GSM MT calls in MNCC format, going through themwi-mncc | 34 and turns them into GSM MT calls in MNCC format, going through themwi-mncc |
35 and ultimately to OsmoMSC. | 35 and ultimately to OsmoMSC. |
36 | 36 |
37 * These inbound calls per the previous bullet point also include fully working | 37 * Our outbound call gateway is themwi-sip-out, serving as a counterpart to |
38 voice path, with our themwi-mgw transcoding the two RTP streams (one in each | 38 themwi-sip-in. All GSM MO calls are initially handled by themwi-mncc, which |
39 direction) between the original GSM 06.10 codec on the GSM side and G.711 | 39 decides (based on the dialed number) whether to switch the call locally |
40 PCMU or PCMA on the PSTN-via-SIP side. This voice call gateway includes | 40 (calling another ThemWi GSM phone) or to pass it to themwi-sip-out. The |
41 working DTMF support: START DTMF and STOP DTMF commands from GSM phones pass | 41 latter process turns outbound calls into SIP and sends them to BulkVS, and we |
42 through OsmoMSC, themwi-mncc and themwi-sip-in to themwi-mgw, and the latter | 42 are successfully able to call any PSTN destination anywhere in the +1 country |
43 process injects in-band DTMF tones into the G.711 RTP stream that is otherwise | 43 code. (themwi-sip-out has provisions for international routes too, but we |
44 generated by transcoding from GSM voice codecs. | 44 haven't set up any yet.) |
45 | |
46 * These inbound and outbound calls per the previous two bullet points also | |
47 include fully working voice path, with our themwi-mgw transcoding the two RTP | |
48 streams (one in each direction) between the original GSM 06.10 codec or | |
49 GSM 06.60 EFR codec on the GSM side and G.711 PCMU or PCMA on the PSTN-via-SIP | |
50 side. This voice call gateway includes working DTMF support: START DTMF and | |
51 STOP DTMF commands from GSM phones pass through OsmoMSC, themwi-mncc and | |
52 themwi-sip-{in,out} to themwi-mgw, and the latter process injects in-band DTMF | |
53 tones into the G.711 RTP stream that is otherwise generated by transcoding | |
54 from GSM voice codecs. | |
45 | 55 |
46 The following functionality remains to be implemented: | 56 The following functionality remains to be implemented: |
47 | 57 |
48 * As a counterpart to themwi-sip-in, there will be another process named | 58 * The only GSM codecs currently supported are the original FR (GSM 06.10) and |
49 themwi-sip-out that will serve as a gateway for outbound calls, going from | 59 EFR (GSM 06.60). At some point it would be nice to add support for AMR and |
50 GSM MO MNCC to outside PSTN via SIP. The outbound SIP call functional part | 60 maybe even HR1, purely for demonstration of its poor voice quality. |
51 is already implemented in test prototype form in sip-manual-out. | |
52 | 61 |
53 * Right now themwi-mgw supports only the original FR1 codec (GSM 06.10) on the | 62 * The Presiding High Priestess of Themyscira Wireless has a desire to implement |
54 GSM side; the Mother's desire is to also support EFR codec as a high priority, | 63 in-band GSM 08.62 TFO inside our G.711 RTP interface to outside PSTN. Right |
55 and maybe some time later AMR as a lower priority. | 64 now if a call is connected between a ThemWi GSM phone on one end and another |
65 GSM phone on some other network anywhere else in the world, an undesirable | |
66 tandem transcoding takes place: two lossy GSM speech codecs run in tandem, | |
67 one on the ThemWi GSM leg of the call and another on the distant GSM network's | |
68 leg, with a G.711 64 kbps connection in between. If we implement GSM 08.62 | |
69 TFO inside our G.711 RTP interface which we present to the outside world, we | |
70 shall achieve TFO calls at least between ThemWi and any other community GSM | |
71 networks using our software, and maybe even between ThemWi and some legacy | |
72 commercial GSM networks, if any still remain in parts of the world that are | |
73 accessible to us. | |
74 | |
75 * No work whatsoever has been done on SMS as of yet - all of our ThemWi work so | |
76 far has been in support of voice call functionality only. Adding SMS support | |
77 will require 3 major areas of work: | |
78 | |
79 1) We will need to climb the learning curve and assess the current state of SMS | |
80 support in Osmocom CNI software. | |
81 | |
82 2) We'll have to find a USA PSTN connectivity provider who supports P2P SMS: if | |
83 we are operating a GSM network and providing mobile phone service to human | |
84 end users, any SMS usage in such scenario MUST be classified as P2P rather | |
85 than A2P. However, gaining access to P2P SMS in USA telecom environment is | |
86 currently very difficult (every major provider forcibly imposes A2P | |
87 misclassification with no option of P2P correct classification), in stark | |
88 contrast to very easy and cheap access to voice PSTN with NANP phone numbers. | |
89 | |
90 3) When and if we find that necessary USA PSTN connectivity provider with P2P | |
91 SMS provision, we will need to implement whatever software will be needed to | |
92 interconnect it with an OsmoCNI GSM network, likely a custom SMSC. | |
56 | 93 |
57 Differences from osmo-sip-connector | 94 Differences from osmo-sip-connector |
58 ----------------------------------- | 95 ----------------------------------- |
59 | 96 |
60 In the Osmocom community, the "standard" (or generally accepted) way to connect | 97 In the Osmocom community, the "standard" (or generally accepted) way to connect |