# HG changeset patch
# User Mychaela Falconia <falcon@freecalypso.org>
# Date 1710189909 28800
# Node ID 372be50488d63b694821e39acdb63ccea9583897
# Parent  2cdbd574bba6353d5981eeefa35b5132db96c45b
add README

diff -r 2cdbd574bba6 -r 372be50488d6 README
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/README	Mon Mar 11 12:45:09 2024 -0800
@@ -0,0 +1,27 @@
+This repository contains a collection of programs that need to be run under
+the following somewhat constraining conditions:
+
+1) The developer-operator running these programs needs to be someone with an
+   active account at BulkVS, AnveoDirect or some other similar provider of SIP
+   trunk access to PSTN;
+
+2) The test programs in this collection need to be run on a machine with a
+   direct, non-NAT'ed connection to public Internet, one whose static & public
+   IP address is whitelisted with the service provider from the previous bullet
+   point as a customer host authorized to send outgoing calls to PSTN on the
+   customer's account.
+
+Each of sipout-test-* utilities in this collection is a command line tool that
+generates an outbound call on PSTN (G.711 codecs only, either PCMU or PCMA) via
+one of the just-described SIP trunk providers.  The breakdown is as follows:
+
+* sipout-test-voice is intended for testing voice calls and the in-band signals
+  for GSM 08.62 or 3GPP TS 28.062 TFO;
+
+* sipout-test-fsk generates a 300 baud FSK modem call, either Bell103 or V.21;
+
+* sipout-test-v22 generates a V.22(bis) modem call at either 1200 or 2400 bps.
+
+These programs use SpanDSP library for V-series modulations (required for
+compilation) and themwi-rtp-mgr (required at run time) to get a properly aligned
+(even/odd) RTP+RTCP port pair for talking to PSTN-via-SIP.