--- a/README	Tue Sep 20 23:20:50 2022 -0800
+++ b/README	Wed Sep 21 07:58:16 2022 -0800
@@ -3,22 +3,11 @@
 by Mother Mychaela of FreeCalypso at a semi-urban/semi-rural location in
 Southern California, USA; this GSM network is operated with Osmocom CNI software
 components, all running on a single Slackware Linux server.  ThemWi system sw
-is going to be a suite of daemon processes and command line tools that run on
-the same machine as all those Osmocom sw components and provide some additional
+is a suite of daemon processes and command line tools that run on the same
+machine as all those Osmocom sw components and provide some additional
 functionality that is not provided "out of the box" by Osmocom, most important
 of which is outside connectivity to USA PSTN.
 
-Right now we have a themwi-mncc daemon process that connects to OsmoMSC via the
-MNCC socket interface provided by the latter and takes the place of OsmoMSC's
-mncc_builtin.  themwi-mncc switches local calls (from one GSM subscriber to
-another) just like mncc_builtin, but it also provides a hook (mtcall_socket)
-for routing externally originated calls to GSM (which then become MT calls),
-and it will later have a similar interface for routing MO calls to the outside
-world.  Additional daemon processes that will interface with USA PSTN via SIP
-(one process accepting SIP INVITEs from bulkvs.com servers, turning them into
-MNCC and sending the calls toward GSM, and another process going the other way)
-remain to be implemented.
-
 We are currently experimenting with using bulkvs.com as our USA PSTN
 connectivity provider.  Like most low-cost PSTN connectivity providers, they
 provide the interface to PSTN in the form of a SIP trunk - while I would
@@ -27,19 +16,116 @@
 Our current status is:
 
 * We have already obtained a block of USA phone numbers (NANP, chosen numbers
-  from an exchange area local to us) from bulkvs.com;
+  from an exchange area local to us) from BulkVS.
+
+* These BulkVS-sourced real NANP numbers have been entered as MSISDNs into
+  OsmoHLR records for our test SIMs operating on ThemWi GSM.
+
+* We can successfully dial calls from one ThemWi GSM phone to another, with our
+  themwi-mncc switch understanding all dialing formats that are considered
+  standard for cellular phone networks in USA (full international, or 11 digits
+  starting with '1' but no '+', or 10 digits only), as well as our own non-
+  standard shorthand dialing method with only 4 digits.
+
+* Whenever someone dials one of our NANP numbers from the outside world, BulkVS
+  servers send UDP SIP INVITE packets to our server.  Our inbound call gateway
+  process is themwi-sip-in; this daemon process listens on UDP port 5060,
+  accepts SIP calls from BulkVS (ultimately coming from global worldwide PSTN)
+  and turns them into GSM MT calls in MNCC format, going through themwi-mncc
+  and ultimately to OsmoMSC.
+
+The following functionality remains to be implemented:
+
+* As a counterpart to themwi-sip-in, there will be another process named
+  themwi-sip-out that will serve as a gateway for outbound calls, going from
+  GSM MO MNCC to outside PSTN via SIP.  The outbound SIP call functional part
+  is already implemented in test prototype form in sip-manual-out.
 
-* These bulkvs-sourced real NANP numbers have been entered as MSISDNs into
-  OsmoHLR records for our test SIMs operating on ThemWi GSM;
+* themwi-mgw will be our transcoding RTP bridge, speaking GSM codecs (FR and
+  EFR are currently of most interest) on the side toward Osmocom components and
+  G.711 (PCMU or PCMA) on the PSTN side.  Right now themwi-mgw is a working
+  skeleton that allocates endpoints with RTP & RTCP UDP port pairs, but doesn't
+  pass any traffic yet.
+
+Differences from osmo-sip-connector
+-----------------------------------
+
+In the Osmocom community, the "standard" (or generally accepted) way to connect
+a GSM voice network to the outside world is via osmo-sip-connector, an Osmocom
+process that connects to OsmoMSC's MNCC socket on one end and talks SIP on the
+other end.  Our combination of themwi-mncc, themwi-sip-in and themwi-sip-out
+effectively takes the place of osmo-sip-connector.  Here are the principal ways
+in which our solution differs from osmo-sip-connector:
+
+* o-s-c is designed to connect to a local instance of a SIP PBX such as Asterisk
+  or FreeSWITCH, as opposed to interfacing directly to an outside SIP trunk
+  from/to a PSTN-via-SIP connectivity provider.  themwi-system-sw is different
+  in this regard: we do NOT use Asterisk or FreeSWITCH or any other similar
+  software of "spaceship" complexity, instead our themwi-sip-in and
+  themwi-sip-out processes interface directly to our PSTN-via-SIP connectivity
+  provider.
+
+* o-s-c has no internal call switching function for calls from one local GSM
+  phone to another, instead such switching is punted to the required Asterisk
+  or FreeSWITCH etc.  With o-s-c, the calling phone's MO call is converted to
+  SIP, then Asterisk or other PBX hairpins it back to o-s-c, and then o-s-c
+  handles the destination call leg as a separate conversion from SIP back to
+  GSM MNCC.  In our solution such local calls are switched internally inside
+  themwi-mncc, staying native within GSM MNCC land and never turning into SIP.
 
-* We can successfully dial calls from one ThemWi GSM phone to another, with
-  themwi-mncc understanding all dialing formats that are considered standard
-  for cellular phone networks in USA (full international, or 11 digits starting
-  with '1' but no '+', or 10 digits only), as well as our own non-standard
-  shorthand dialing method with only 4 digits;
+* o-s-c is based on Sofia-SIP, which in turn uses glib - a very unpleasant
+  dependency in this Mother's opinion.  In contrast, our implementation of SIP
+  is 100% from scratch, written in plain C in the traditional Falconian coding
+  style.
+
+The need for RTP voice transcoding
+----------------------------------
+
+In the context of GSM voice codecs, the term "transcoding" is used in two
+significantly different meanings:
+
+* Transcoding from one lossy GSM codec to another effectively constitutes two
+  lossy speech codecs running in tandem, and is a highly undesirable condition.
+
+* Running a single GSM codec (not two in tandem), decoding from GSM to G.711 in
+  one direction and encoding from G.711 in the other direction, is a standard
+  required function for traditional voice gateways between GSM and PSTN.
+
+In themwi-system-sw, we need to do transcoding in the second sense above.
+BulkVS SIP call interface to PSTN does not support any of GSM codecs, they only
+support G.711 and G.729, and the same situation is expected to hold with other
+PSTN-via-SIP connectivity providers.  We certainly don't want to use G.729 - we
+don't want to run two lossy speech codecs in tandem, first GSM and then G.729 -
+hence the only codecs we speak on the PSTN-via-SIP side of our gateway are PCMU
+and PCMA.  Therefore, we need to perform RTP transcoding in our themwi-mgw,
+very similar to a traditional GSM TRAU.
 
-* Whenever someone dials one of our NANP numbers from the outside world, bulkvs
-  servers send UDP SIP INVITE packets to our server, and by dumping them with
-  our sip-udp-dump utility, we can see exactly what we have to work with;
+On the GSM side, the two codecs of most interest to us at the present time are
+the original FR and EFR - hence they will be the first to be supported.  Note
+the big difference from other Osmocom-using GSM community networks which
+typically prefer or even strictly require AMR instead!  Our reasons for focusing
+on FR and EFR instead of AMR are:
+
+* Our OsmoBSC time slot configuration is full rate channels only, no half rate
+  channels.  HR channels are needed only for greater capacity of simultaneous
+  calls, but with the total number of people *on the planet* who actively want
+  GSM/2G as opposed to LTE or 5G being no more than maybe 10, the thought of
+  exceeding the limit of 6 simultaneous call legs per cell (meaning 6 separate
+  GSM phone handsets talking *at the same time*) is preposterous.
 
-* The rest remains to be implemented.
+* Without any HR channels in OsmoBSC config, AMR means AMR-FR specifically, not
+  AMR-HR.  The highest level of AMR-FR is identical with EFR - thus if we
+  support EFR, do we really need AMR?
+
+* The whole point of Themyscira Wireless is to provide service to *vintage*
+  mobile phones.  Our current collection of vintage phones includes models that
+  only support FR1 and EFR (Ericsson I888, Nokia 5190) and Calypso C05 which
+  supports FR1, EFR and HR1, but not AMR.
+
+* EFR is desirable because it gives better voice quality than FR1, but we must
+  support FR1 too, so we can serve the very oldest of phones which support only
+  FR1 and nothing else.
+
+Voice codec restriction (forcing GSM phones to use EFR instead of AMR, or
+forcing all the way down to FR1) is done in OsmoBSC config, with codec-list
+setting under 'msc 0'.