FreeCalypso > hg > sipout-test-utils

annotate README @ 16:4f81b959a5f5

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sipout-test-voice: implement PCMU GSM uplink catcher
author Mychaela Falconia <falcon@freecalypso.org>
date Mon, 13 May 2024 22:10:25 -0800
parents 372be50488d6
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1 This repository contains a collection of programs that need to be run under
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2 the following somewhat constraining conditions:
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3
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4 1) The developer-operator running these programs needs to be someone with an
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5 active account at BulkVS, AnveoDirect or some other similar provider of SIP
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6 trunk access to PSTN;
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8 2) The test programs in this collection need to be run on a machine with a
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9 direct, non-NAT'ed connection to public Internet, one whose static & public
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10 IP address is whitelisted with the service provider from the previous bullet
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11 point as a customer host authorized to send outgoing calls to PSTN on the
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12 customer's account.
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14 Each of sipout-test-* utilities in this collection is a command line tool that
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15 generates an outbound call on PSTN (G.711 codecs only, either PCMU or PCMA) via
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16 one of the just-described SIP trunk providers. The breakdown is as follows:
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18 * sipout-test-voice is intended for testing voice calls and the in-band signals
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19 for GSM 08.62 or 3GPP TS 28.062 TFO;
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21 * sipout-test-fsk generates a 300 baud FSK modem call, either Bell103 or V.21;
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23 * sipout-test-v22 generates a V.22(bis) modem call at either 1200 or 2400 bps.
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25 These programs use SpanDSP library for V-series modulations (required for
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26 compilation) and themwi-rtp-mgr (required at run time) to get a properly aligned
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27 (even/odd) RTP+RTCP port pair for talking to PSTN-via-SIP.