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sipout-test-voice: implement play with offset
author Mychaela Falconia <falcon@freecalypso.org>
date Sat, 11 May 2024 22:27:07 -0800
parents 372be50488d6
children
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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.